[asterisk-users] Today on VUC, Dan York on Google Voice + SIP
Hi, Today at 12 Non EDT, Dan York will be with us to talk about the recent on and off moments of Google Voice SIP URI calling. Like Skype + Asterisk (or any SIP), Google Voice and SIP compose the other shoe waiting to drop. We're following this with interest. So GV turned on SIP URI and then a few days later, turned it off. Why? Did the geeks (like us) jump on this too quickly or too heavily? Dan's Disruptive Telephony site is a reference in the field and we'll likely be talking about other news of interest as well. We'd love to have you join on on our call with this week's guest OnSIP.com by connecting via these technologies: SIP:200...@login.zipdx.com - use g722 if you have it Skype:vuc.me PSTN: +1 567 252 2286 iNum: +883 5100 123 94882 Text backchat on #vuc channel of Freenode.net - use http://vuc.me/irc if you don't have a client You can also talk to us on Twitter @voipusers or the hashtag #vuc More info: http://vuc.me Hope to hear you soon. /r -- _ -- 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] Asterisk 1.6.2.10 CDR custom added field
On Thursday 24 March 2011 04:50:48 Jonas Kellens wrote: On 03/24/2011 10:45 AM, Rizwan Hisham wrote: You have to use adaptive cdr for this functionality. In 1.8 the conf file for adaptive cdr is cdr_adaptive_odbc.conf. The sample conf file should tell you everything. If you are using some other cdr engine then you will have to jump into the code of asterisk to make it log the item you want, which includes creating an extra variable in the cdr data struction, creating a function to set/get its value from dialplan, and then changing the sql command to include the extra variable for insertion into DB. I thought it was possible in asterisk 1.6.2 to add extra mysql-fields ?? In asterisk 1.4 you just have one 'userfield', but in 1.6.2 it is possible to add custom fields... I just don't know how. This is what the wiki (http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql) tells : /Module now permits arbitrary columns to be created and populated, just like cdr_adaptive_odbc, simply by adding the column to the table and defining the corresponding CDR() variable/ Where is the information on this ? Same as always, in the configs/ directory of addons 1.6.2. The sample configuration file contains common examples of the added functionality. Also, there's a note on it in UPGRADE.txt, in the root directory of addons 1.6.2. If you have any further questions, you're welcome to ask this list. -- Tilghman -- _ -- 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] What is the most stable version of asterisk?
On Thursday 24 March 2011 11:38:54 Gordon Henderson wrote: On Wed, 23 Mar 2011, Douglas Mortensen wrote: 1.2? 1.4? 1.6? 1.8? 1.2 has been the most stable version for me. Same setups with 1.4 +DAHDI has never been as stable with random crashes and re-starts - however they're not predictable and sometimes months apart. I had one instance of 1.2 run for over a year without a hiccup. I've not even thought about 1.8 yet. There is an inherent danger in running 1.2 code at this point, however. Any security issue that applies to 1.2 won't be patched by the Asterisk team, now that it has passed out of security maintenance mode. You'll need to watch for future vulnerability reports, keeping in mind that some vulnerabilities will only apply to Asterisk 1.2, not later versions (in which case Digium _may_ silently ignore the reports), and you may need to patch those manually. Depending upon your setup, this may or may not be a big concern, but you should at least be aware of it. If your Asterisk 1.2 box is public-facing, this is a potential risk that you should mitigate. -- Tilghman -- _ -- 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] Filtering on from caller id
That sounds good but, i would like it the other way arround. I have over 90 extensions that are NOT allowed to use the trunk, and 2 that are.. So blacklisting everything will take for ever ;D. On Thu, Mar 24, 2011 at 10:01 PM, Danny Nicholas da...@debsinc.com wrote: Just use “Ex-girlfriend” logic on your dial command to zap 103 when he tries to use the trunk. For each dial command in your dialplan that addresses the trunk (let’s call it DAHDI/1 for brevity), duplicate the line like this: Existing: Exten = _X.,1,Dial(DAHDI/1,w,5551212) New: Exten = _X,1,noop(everybody but 103 dials) Exten = _X./103,n,hangup Exten = _X.,n,Dial(DAHDI/1,w,5551212) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Peter den Hartog *Sent:* Thursday, March 24, 2011 3:45 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Filtering on from caller id Hmm are those extensions then still able to call each other? I'll put it more clear, I have: 100, 101, 103. A Trunk. 100 101 are allowed to call over the trunk. but 103, is only allowed to call with 100 101, of course 100 101 should still be able to call 103. On Thu, Mar 24, 2011 at 5:14 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Thursday 24 Mar 2011, Peter den Hartog wrote: I would like to use the from caller id, to allow calls yes or no. 101, and 111 should be allowed to use the Trunk, the rest of the phones are not. Is this even possible? So if the from caller id is 101 or 111, then allow the call, otherwise hangup. Would it not be simpler just to put those extensions into a separate context, which allows trunk calls? -- AJS Answers come *after* questions. -- _ -- 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
Re: [asterisk-users] Filtering on from caller id
So make a whitelist What I do is create a outbound route with the allowed cid and then have another route which goes to a not allowed recording which catches all other caller Id's -Original Message- From: Peter den Hartog peterdenhar...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 25 Mar 2011 09:14:45 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Filtering on from caller id -- _ -- 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
Re: [asterisk-users] Fwd: asking for some help
On Thursday 24 March 2011 12:02:38 vip killa wrote: If you are new to VoIP, you are better off learning FreeSWITCH And if you're new to analog recordings, you're better off purchasing Sony BetaMax. How is your BetaMax deck, btw? -- Tilghman -- _ -- 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] using ${EXTEN} with waitexten
Hi
Using ${EXTEN:0:3}
will only return the first 3 digits entered
Ish
On Wed, 2011-03-23 at 16:27 -0400, Eddie Mikell wrote:
All:
Some of the people who dial into to our system will press the pound key
when entering an extension for the directory key. When waitexten gets
that, I get an error messages as, for example 123# doesn't match any
extension.
I was going to use ${EXTEN} to just use the first three numbers, but I'm
not sure how to use this with WaitExten.
so I have
exten = 4349701010,1,Answer()
exten = 4349701010,2,ringing
exten = 4349701010,3,wait(8)
exten = 4349701010,4,Background(asterisk-recording)
exten = 4349701010,5,WaitExten(9,m)
exten = 4349701010,6,Dial(SIP/100SIP/123SIP/132SIP/134SIP/149,20)
exten = 4349701010,7,VoiceMail(100@default,u)
exten = 4349701010,8,Playback(vm-goodbye)
exten = 4349701010,9,Hangup()
Where could I check for the extra # keystroke?
Thanks for your help.
eddie
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Re: [asterisk-users] Sox and bad quality when converting to 8 kHz
On Thu, 2011-03-24 at 21:58 +0100, Thomas Winter wrote:
Hi list,
I have an 44100 Hz file with human voice, stereo with 16Bit.
When convertig this to 8 kHz, mono I loose a lot of quality and have
some ground noise. I tried several sox options but without success.
Can somebody help
best regards Thomas
The best results I have had have been by using the following
mpg123 -q -w ${TEMP} ${INPUT}
sox ${TEMP} -c 1 -s -r 8000 ${OUTPUT}
Regards
Ish
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Software Developer
PackNet Ltd
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Re: [asterisk-users] Problems Extension with a Call In on Asterisk 1.6
Hi Olivier,
here is solutions for your situation , ideally you need to talk with
Provider and they can set SIP URI
for given DID numbre , but that can be solved by dial-plan like this.
exten = _003318364,1,Set(foo=${SIP_HEADER(To)})
exten = _003318364,n,Set(cut1=${CUT(foo,:,2)})
exten = _003318364,n,Set(CLI=${CUT(cut1,,1)})
exten = _003318364,n,Set(toexten=${CUT(CLI,@,1)})
exten = _003318364,n,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten = _003318364,n,ExecIf($[${toexten} =
81169]?Dial(SIP/204,180,rt):Noop(${toexten}))
exten = _003318364,n,ExecIf($[${EXTEN} =
003318364]?Dial(SIP/203,180,rt):Noop(${toexten}))
On Thu, Mar 24, 2011 at 11:13 AM, Olivier CALVANO o.calv...@gmail.comwrote:
Hi
Anyone know a solution at my problems ?
Thanks
Olivier
2011/3/23 Olivier CALVANO o.calv...@gmail.com:
Hi
I request your help because i don't have actually a solution at my
problems.
I have a Asterisk Server in 1.6
Connected at a SIP Provider
This provider supply me 2 numbers:
003318364 (official number)
081169 (Nddi Number)
When i receive a call on the 081169, he don't use
the extension. He use the 003318364 extension.
SIP Debug:
--- SIP read from UDP://91.121.xxx.xxx:5060 ---
INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0
Allow: UPDATE,REFER,INFO
Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
Contact: sip:91.121.xxx.xxx:5060
Content-Type: application/sdp
CSeq: 1602837515 INVITE
From: 033426aa
sip:033426aaa...@sip.myoperator.net
;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60
Max-Forwards: 30
P-Preferred-Identity: sip:033426aaa...@sip.myoperator.net;user=phone
To: sip:081169x...@91.121.xxx.xxx;user=phone
User-Agent: Cirpack/v4.42s (gw_sip)
Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2
Content-Length: 481
v=0
o=cp10 130085910854 130085910854 IN IP4 10.7.1.121
s=SIP Call
c=IN IP4 91.121.bbb.bbb
t=0 0
m=audio 36146 RTP/AVP 18 4 0 8 125 111 101
b=AS:21
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000/1
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:125 CLEARMODE/8000/1
a=rtpmap:111 iLBC/8000/1
a=fmtp:111 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=sqn:0
a=cdsc: 1 image udptl t38
-
--- (13 headers 22 lines) ---
Sending to 91.121.xxx.xxx : 5060 (no NAT)
Using INVITE request as basis request -
04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
Found peer 'Myoperator' for '033426aa' from 91.121.xxx.xxx:5060
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 125
Found RTP audio format 111
Found RTP audio format 101
Peer audio RTP is at port 91.121.bbb.bbb:36146
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CLEARMODE for ID 125
Found audio description format iLBC for ID 111
Found audio description format telephone-event for ID 101
Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d
(g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing),
combined - 0x109 (g723|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 91.121.bbb.bbb:36146
Looking for 003318364 in Appels-Entrants (domain 78.41.xxx.xxx)
--- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx
From: 033426aa
sip:033426aaa...@sip.myoperator.net
;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60
To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a
Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
CSeq: 1602837515 INVITE
Server: Asterisk PBX 1.6.1.8
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
[Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527
handle_request_invite: Call from '0033459aa' to extension
'003318364' rejected because extension not found.
Scheduling destruction of SIP dialog
'04459-nk-5fa6f8a0-18641f...@sip.myoperator.net' in 6400 ms (Method:
INVITE)
--- SIP read from UDP://91.121.xxx.xxx:5060 ---
ACK sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0
Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
Contact: sip:91.121.xxx.xxx:5060
CSeq: 1602837515 ACK
From: 033426aa
sip:033426aaa...@sip.myoperator.net
;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60
Max-Forwards: 30
To:
Re: [asterisk-users] Filtering on from caller id
That sounds good, do you have a example of that? On Fri, Mar 25, 2011 at 9:24 AM, isr...@gmail.com wrote: So make a whitelist What I do is create a outbound route with the allowed cid and then have another route which goes to a not allowed recording which catches all other caller Id's -Original Message- From: Peter den Hartog peterdenhar...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 25 Mar 2011 09:14:45 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Filtering on from caller id -- _ -- 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
Re: [asterisk-users] Asterisk 1.6.2.10 CDR custom added field
On 03/25/2011 08:19 AM, Tilghman Lesher wrote: On Thursday 24 March 2011 04:50:48 Jonas Kellens wrote: On 03/24/2011 10:45 AM, Rizwan Hisham wrote: You have to use adaptive cdr for this functionality. In 1.8 the conf file for adaptive cdr is cdr_adaptive_odbc.conf. The sample conf file should tell you everything. If you are using some other cdr engine then you will have to jump into the code of asterisk to make it log the item you want, which includes creating an extra variable in the cdr data struction, creating a function to set/get its value from dialplan, and then changing the sql command to include the extra variable for insertion into DB. I thought it was possible in asterisk 1.6.2 to add extra mysql-fields ?? In asterisk 1.4 you just have one 'userfield', but in 1.6.2 it is possible to add custom fields... I just don't know how. This is what the wiki (http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql) tells : /Module now permits arbitrary columns to be created and populated, just like cdr_adaptive_odbc, simply by adding the column to the table and defining the corresponding CDR() variable/ Where is the information on this ? Same as always, in the configs/ directory of addons 1.6.2. The sample configuration file contains common examples of the added functionality. Also, there's a note on it in UPGRADE.txt, in the root directory of addons 1.6.2. If you have any further questions, you're welcome to ask this list. All I can find is the following : ; You may also configure the field names used in the CDR table. ; [columns] ;static value = column ;alias cdrvar = column alias start = calldate alias callerid = clid ;alias src = src ;alias dst = dst ;alias dcontext = dcontext ;alias channel = channel ;alias dstchannel = dstchannel ;alias lastapp = lastapp ;alias lastdata = lastdata ;alias duration = duration ;alias billsec = billsec ;alias disposition = disposition ;alias amaflags = amaflags ;alias accountcode = accountcode ;alias userfield = userfield ;alias uniqueid = uniqueid But this is not explained... So please can you confirm how I think it should work : In my dialplan I have : /exten = 600,n,Set(CDR(mycolumn)=myvalue)/ So I should add the following to cdr_mysql.conf : /[columns] static mycolumn = mycolumn/ And if I want this in my dialplan : /exten = 600,n,Set(CDR(anothercolumn)=anothervalue)/ then I first need to add to cdr_mysql.conf : /static anothercolumn = anothercolumn/ Can you confirm ? Kind regards, Jonas. -- _ -- 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] Filtering on from caller id
One extra line to change blacklist to whitelist Exten = _X,1,noop(everybody but 103 dials) Exten = _X./100,n,Dial(DAHDI/1,w,5551212) Exten = _X./101,n,Dial(DAHDI/1,w,5551212) Exten = _X.,n,hangup _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter den Hartog Sent: Friday, March 25, 2011 3:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Filtering on from caller id That sounds good but, i would like it the other way arround. I have over 90 extensions that are NOT allowed to use the trunk, and 2 that are.. So blacklisting everything will take for ever ;D. On Thu, Mar 24, 2011 at 10:01 PM, Danny Nicholas da...@debsinc.com wrote: Just use Ex-girlfriend logic on your dial command to zap 103 when he tries to use the trunk. For each dial command in your dialplan that addresses the trunk (let's call it DAHDI/1 for brevity), duplicate the line like this: Existing: Exten = _X.,1,Dial(DAHDI/1,w,5551212) New: Exten = _X,1,noop(everybody but 103 dials) Exten = _X./103,n,hangup Exten = _X.,n,Dial(DAHDI/1,w,5551212) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter den Hartog Sent: Thursday, March 24, 2011 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Filtering on from caller id Hmm are those extensions then still able to call each other? I'll put it more clear, I have: 100, 101, 103. A Trunk. 100 101 are allowed to call over the trunk. but 103, is only allowed to call with 100 101, of course 100 101 should still be able to call 103. On Thu, Mar 24, 2011 at 5:14 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Thursday 24 Mar 2011, Peter den Hartog wrote: I would like to use the from caller id, to allow calls yes or no. 101, and 111 should be allowed to use the Trunk, the rest of the phones are not. Is this even possible? So if the from caller id is 101 or 111, then allow the call, otherwise hangup. Would it not be simpler just to put those extensions into a separate context, which allows trunk calls? -- AJS Answers come *after* questions. -- _ -- 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
Re: [asterisk-users] Sox and bad quality when converting to 8 kHz
On 03/25/2011 04:58 AM, Thomas Winter wrote: Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help best regards Thomas You really need to remove the bass end of the spectrum before downsampling to 8k/s. uLaw/ALaw sound pretty muddy and horrible if you don't do that, and the other common 8k/s codecs don't sound any better. Jean-Marc Valin wrote a little filtering utility for this purpose. Steve -- _ -- 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
[asterisk-users] asterisk 1.8 question
In 1.4 there was core show channels concise This seems to be gone from 1.8. When I am using the AMI interface to get a listing of all channels my listing names are cut short. SIP/devcentos5x64_to notice above. In 1.4 it would have given me SIP/devcentos5x64_to_am2mm How in 1.8 do I get the FULL listing of the channels. THanks, Jerry -- _ -- 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] What is the most stable version of asterisk?
Based on the following URL, it seems that CallWeaver may not still be an active project?? http://www.callweaver.org/blog/20 From a security standpoint, I would usually expect it is safer to be with an active project, than a dead one. Otherwise who is going to patch vulnerabilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stot...@totarotechnologies.com] Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the most stable version of asterisk? On Wed, Mar 23, 2011 at 12:58 PM, Douglas Mortensen d...@impalanetworks.com wrote: 1.2? 1.4? 1.6? 1.8? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 Callweaver? http://www.voip-info.org/wiki/view/CallWeaver. I believe they forked somewhere in the 1.2 release. Many features ahead of Asterisk. Although I didn't see anything on FreeSwitch stating anything anything about deadlocking, I know that was one of the main reasons for BKW, as seasoned asterisk developer and folks to start from scratch. That and the hybrid dual license in Asterisk. http://www.sofaswitch.org/docs/How%20does%20FreeSWITCH%20compare%20to%20Asterisk.pdf Read the whole piece. I know it isn't Asterisk but BKW who contributed and I believe is still helping Asterisk Besides, I feel that FreeSwitch is the most stable. I like 1.2 so I went with Callweaver for many installations. Thanks, Steve Totaro -- _ -- 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] What is the most stable version of asterisk?
Don't have to be a developer to be a patcher, but it helps ... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Friday, March 25, 2011 9:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the most stable version of asterisk? Based on the following URL, it seems that CallWeaver may not still be an active project?? http://www.callweaver.org/blog/20 From a security standpoint, I would usually expect it is safer to be with an active project, than a dead one. Otherwise who is going to patch vulnerabilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stot...@totarotechnologies.com] Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the most stable version of asterisk? On Wed, Mar 23, 2011 at 12:58 PM, Douglas Mortensen d...@impalanetworks.com wrote: 1.2? 1.4? 1.6? 1.8? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 Callweaver? http://www.voip-info.org/wiki/view/CallWeaver. I believe they forked somewhere in the 1.2 release. Many features ahead of Asterisk. Although I didn't see anything on FreeSwitch stating anything anything about deadlocking, I know that was one of the main reasons for BKW, as seasoned asterisk developer and folks to start from scratch. That and the hybrid dual license in Asterisk. http://www.sofaswitch.org/docs/How%20does%20FreeSWITCH%20compare%20to%20Aste risk.pdf Read the whole piece. I know it isn't Asterisk but BKW who contributed and I believe is still helping Asterisk Besides, I feel that FreeSwitch is the most stable. I like 1.2 so I went with Callweaver for many installations. Thanks, Steve Totaro -- _ -- 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
Re: [asterisk-users] What is the most stable version of asterisk?
I have been somewhat interested in FreeSwitch in the past, but I am mostly interested in Asterisk. That's why I asked about stability of asterisk versions. Maybe some other time I'll look deeper into FreeSwitch. Thanks. And thanks everyone for the feedback. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stot...@totarotechnologies.com] Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the most stable version of asterisk? On Wed, Mar 23, 2011 at 12:58 PM, Douglas Mortensen d...@impalanetworks.commailto:d...@impalanetworks.com wrote: 1.2? 1.4? 1.6? 1.8? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.comhttp://www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 Callweaver? http://www.voip-info.org/wiki/view/CallWeaver. I believe they forked somewhere in the 1.2 release. Many features ahead of Asterisk. Although I didn't see anything on FreeSwitch stating anything anything about deadlocking, I know that was one of the main reasons for BKW, as seasoned asterisk developer and folks to start from scratch. That and the hybrid dual license in Asterisk. http://www.sofaswitch.org/docs/How%20does%20FreeSWITCH%20compare%20to%20Asterisk.pdf Read the whole piece. I know it isn't Asterisk but BKW who contributed and I believe is still helping Asterisk Besides, I feel that FreeSwitch is the most stable. I like 1.2 so I went with Callweaver for many installations. Thanks, Steve Totaro -- _ -- 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] What is the most stable version of asterisk?
Do you have the same ratio of deployments using 1.4 as you do with 1.2? What about 1.6 or 1.8? I simply question how accurate a comparison can be made when one is comparing 50 1.2 deployments to 5 1.4 deployments? I'm sure it says something, and I do appreciate the feedback. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . -Original Message- From: C F [mailto:shma...@gmail.com] Sent: Thursday, March 24, 2011 8:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the most stable version of asterisk? I use mainly 1.2 with great success, mostly restarts are due to power outages. I recently started to upgrade to 1.4, so far so good. Too early to say, the longest running 1.4 is only 234 days. While I have had 900+ days with 1.2 -- _ -- 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] What is the most stable version of asterisk?
Hi Doug! I use Asterisk 1.4 and 1.8, I can easily see that Asterisk 1.8 works better than 1.4. Everything on Asterisk 1.8 seems better. Best regards, From: d...@impalanetworks.com To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 08:32:04 -0600 Subject: Re: [asterisk-users] What is the most stable version of asterisk? Do you have the same ratio of deployments using 1.4 as you do with 1.2? What about 1.6 or 1.8? I simply question how accurate a comparison can be made when one is comparing 50 1.2 deployments to 5 1.4 deployments? I'm sure it says something, and I do appreciate the feedback. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . -Original Message- From: C F [mailto:shma...@gmail.com] Sent: Thursday, March 24, 2011 8:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the most stable version of asterisk? I use mainly 1.2 with great success, mostly restarts are due to power outages. I recently started to upgrade to 1.4, so far so good. Too early to say, the longest running 1.4 is only 234 days. While I have had 900+ days with 1.2 -- _ -- 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
[asterisk-users] Why shouldn't I use 1.8?
Now that we've hashed out some thoughts on the most stable version of asterisk, I'd like to ask the question as to why I should NOT use 1.8? What are specific reasons? For instance a few days back I was speaking with James at Rhino Equipment. He said that he has no real data on why I shouldn't use 1.8. They just follow a practice of not jumping on the newest version. But I would like specific reasons why I shouldn't use 1.8 in a production environment if anyone has some? Thanks for your feedback! - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- 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] asterisk 1.8 question
On Fri, Mar 25, 2011 at 9:51 AM, Jerry Geis ge...@pagestation.com wrote: In 1.4 there was core show channels concise This seems to be gone from 1.8. When I am using the AMI interface to get a listing of all channels my listing names are cut short. SIP/devcentos5x64_to notice above. In 1.4 it would have given me SIP/devcentos5x64_to_am2mm How in 1.8 do I get the FULL listing of the channels. I think you should try all three below and see which gives you what you like: core show channels core show channels concise core show channels verbose From my experience, they all work in 1.8, but do give different output. -- HTH, - Bob Beers -- _ -- 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] Filtering on from caller id
Ah, makes sense! Thanks! On Fri, Mar 25, 2011 at 2:09 PM, Danny Nicholas da...@debsinc.com wrote: One extra line to change “blacklist” to “whitelist” Exten = _X,1,noop(everybody but 103 dials) Exten = _X./100,n,Dial(DAHDI/1,w,5551212) Exten = _X./101,n,Dial(DAHDI/1,w,5551212) Exten = _X.,n,hangup -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Peter den Hartog *Sent:* Friday, March 25, 2011 3:15 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Filtering on from caller id That sounds good but, i would like it the other way arround. I have over 90 extensions that are NOT allowed to use the trunk, and 2 that are.. So blacklisting everything will take for ever ;D. On Thu, Mar 24, 2011 at 10:01 PM, Danny Nicholas da...@debsinc.com wrote: Just use “Ex-girlfriend” logic on your dial command to zap 103 when he tries to use the trunk. For each dial command in your dialplan that addresses the trunk (let’s call it DAHDI/1 for brevity), duplicate the line like this: Existing: Exten = _X.,1,Dial(DAHDI/1,w,5551212) New: Exten = _X,1,noop(everybody but 103 dials) Exten = _X./103,n,hangup Exten = _X.,n,Dial(DAHDI/1,w,5551212) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Peter den Hartog *Sent:* Thursday, March 24, 2011 3:45 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Filtering on from caller id Hmm are those extensions then still able to call each other? I'll put it more clear, I have: 100, 101, 103. A Trunk. 100 101 are allowed to call over the trunk. but 103, is only allowed to call with 100 101, of course 100 101 should still be able to call 103. On Thu, Mar 24, 2011 at 5:14 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Thursday 24 Mar 2011, Peter den Hartog wrote: I would like to use the from caller id, to allow calls yes or no. 101, and 111 should be allowed to use the Trunk, the rest of the phones are not. Is this even possible? So if the from caller id is 101 or 111, then allow the call, otherwise hangup. Would it not be simpler just to put those extensions into a separate context, which allows trunk calls? -- AJS Answers come *after* questions. -- _ -- 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
Re: [asterisk-users] Sox and bad quality when converting to 8 kHz
On Fri, 25 Mar 2011, Steve Underwood wrote: You really need to remove the bass end of the spectrum before downsampling to 8k/s. uLaw/ALaw sound pretty muddy and horrible if you don't do that, and the other common 8k/s codecs don't sound any better. Jean-Marc Valin wrote a little filtering utility for this purpose. A link? Casual googling didn't yield anything promising. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Why shouldn't I use 1.8?
On Fri, Mar 25, 2011 at 7:36 AM, Douglas Mortensen d...@impalanetworks.com wrote: But I would like specific reasons why I shouldn't use 1.8 in a production environment if anyone has some? That is a loaded question, in that no two environments are likely to be the same. Some bugs are major issues for 1% of the install base and take time to get merged into the code base. You should read through the open issues for the 1.8 branch and see if there are any show stoppers for your environment. If not, try it in the lab and validate that it works for you. Check out https://issues.asterisk.org For my environment specifically, this issue is currently preventing me from migrating from 1.6.2: - 18818 [patch] Crashing when using local channels and realtime on asterisk There are a lot of benefits to the 1.8 branch (Long term support, Called party id, Multicast RTP, etc) but only you can say if it will work with your configuration in your environment. -Jonathan -- _ -- 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] asterisk 1.8 question
From: Bob Beers bob.be...@gmail.com Sent: Friday, March 25, 2011 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk 1.8 question On Fri, Mar 25, 2011 at 9:51 AM, Jerry Geis ge...@pagestation.com wrote: In 1.4 there was core show channels concise This seems to be gone from 1.8. When I am using the AMI interface to get a listing of all channels my listing names are cut short. SIP/devcentos5x64_to notice above. In 1.4 it would have given me SIP/devcentos5x64_to_am2mm How in 1.8 do I get the FULL listing of the channels. I think you should try all three below and see which gives you what you like: core show channels core show channels concise core show channels verbose From my experience, they all work in 1.8, but do give different output. -- HTH, - Bob Beers -- They work for me in 1.8 as well. Bryant -- _ -- 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] Why shouldn't I use 1.8?
From: Jonathan Thurman jonat...@thurmantech.com Sent: Friday, March 25, 2011 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Why shouldn't I use 1.8? On Fri, Mar 25, 2011 at 7:36 AM, Douglas Mortensen d...@impalanetworks.com wrote: But I would like specific reasons why I shouldn't use 1.8 in a production environment if anyone has some? That is a loaded question, in that no two environments are likely to be the same. Some bugs are major issues for 1% of the install base and take time to get merged into the code base. You should read through the open issues for the 1.8 branch and see if there are any show stoppers for your environment. If not, try it in the lab and validate that it works for you. Check out https://issues.asterisk.org For my environment specifically, this issue is currently preventing me from migrating from 1.6.2: - 18818 [patch] Crashing when using local channels and realtime on asterisk There are a lot of benefits to the 1.8 branch (Long term support, Called party id, Multicast RTP, etc) but only you can say if it will work with your configuration in your environment. -Jonathan -- Doug I agree with Jonathan. I have moved all but one of our production switches to 1.8 the only thing holding me back is a minor bug so I have to keep the 1.6.2 box around until that patch is released into the 1.8 branches. When that is done I will no longer be on the 1.6. I have over 98% of our load on the 1.8 switches and we are doing multi tenant pbx hosting and sip trunking. A point of note I just turned down my last 1.4 box 2 weeks ago. It was not because it was not working but because I need more volume and 1.8 on the new hardware meets that need and I get the bonus of not having to support three versions of asterisk now. It is very likely that most of the time I will have at least two versions in production at a time. This is so I can offer the newest features with a stable build and I can offer a more long term support for the customers that the newest features are not as important. Most of my switch hardware has a planned 4 year life span. The better asterisk gets the longer I can stretch that investment. My recommendation is if 1.8 does not have any bugs that are issues for you try 1.8 out of the gate and test, test, test offer feed back from your testing and the bugs will get fixed. I would not spend to much time worrying spend more time doing. Good luck Bryant -- _ -- 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
[asterisk-users] checking dahdi channels
Hi list! Our company is currently using 3 asterisk boxes in 3 locations connected through iax2. Our main office makes and receives many more calls than the other two. I'm looking for a way to check within the dialplan how many channels are in use at the main office so if it reaches a threshold outgoing calls can be iax'ed to one of the satellite locations. Is there a command I could use directly in the dialplan or with the manager interface to get the number of used channels? All locations are running 1.6.2.0 with dahdi 2.2.0.2. thanks -nathan -- _ -- 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] checking dahdi channels
On Fri, 25 Mar 2011, Nathan Pryor wrote: Is there a command I could use directly in the dialplan or with the manager interface to get the number of used channels? Check out the GROUP() and GROUP_COUNT() functions. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Why shouldn't I use 1.8?
Great advice guys. I know it was a loaded question. I appreciate your feedback. Although I'm probably not as much of an asterisk guru as you guys, I tend to agree with your approach. Thanks a lot!! - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Bryant Zimmerman [mailto:brya...@zktech.com] Sent: Friday, March 25, 2011 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why shouldn't I use 1.8? From: Jonathan Thurman jonat...@thurmantech.com Sent: Friday, March 25, 2011 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Why shouldn't I use 1.8? On Fri, Mar 25, 2011 at 7:36 AM, Douglas Mortensen d...@impalanetworks.commailto:d...@impalanetworks.com wrote: But I would like specific reasons why I shouldn't use 1.8 in a production environment if anyone has some? That is a loaded question, in that no two environments are likely to be the same. Some bugs are major issues for 1% of the install base and take time to get merged into the code base. You should read through the open issues for the 1.8 branch and see if there are any show stoppers for your environment. If not, try it in the lab and validate that it works for you. Check out https://issues.asterisk.org For my environment specifically, this issue is currently preventing me from migrating from 1.6.2: - 18818 [patch] Crashing when using local channels and realtime on asterisk There are a lot of benefits to the 1.8 branch (Long term support, Called party id, Multicast RTP, etc) but only you can say if it will work with your configuration in your environment. -Jonathan -- Doug I agree with Jonathan. I have moved all but one of our production switches to 1.8 the only thing holding me back is a minor bug so I have to keep the 1.6.2 box around until that patch is released into the 1.8 branches. When that is done I will no longer be on the 1.6. I have over 98% of our load on the 1.8 switches and we are doing multi tenant pbx hosting and sip trunking. A point of note I just turned down my last 1.4 box 2 weeks ago. It was not because it was not working but because I need more volume and 1.8 on the new hardware meets that need and I get the bonus of not having to support three versions of asterisk now. It is very likely that most of the time I will have at least two versions in production at a time. This is so I can offer the newest features with a stable build and I can offer a more long term support for the customers that the newest features are not as important. Most of my switch hardware has a planned 4 year life span. The better asterisk gets the longer I can stretch that investment. My recommendation is if 1.8 does not have any bugs that are issues for you try 1.8 out of the gate and test, test, test offer feed back from your testing and the bugs will get fixed. I would not spend to much time worrying spend more time doing. Good luck Bryant -- _ -- 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] Why shouldn't I use 1.8?
On 25/03/11 14:36, Douglas Mortensen wrote: Now that we've hashed out some thoughts on the most stable version of asterisk, I'd like to ask the question as to why I should NOT use 1.8? What are specific reasons? For instance a few days back I was speaking with James at Rhino Equipment. He said that he has no real data on why I shouldn't use 1.8. They just follow a practice of not jumping on the newest version. I agree with what Jonathan also said in this thread but that is also a good enough reason on it's own. Data doesn't yet exist to say whether it's stable enough. I like to err on the side of caution with phone systems as they cost lots of money when they go wrong! -- _ -- 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] Why shouldn't I use 1.8?
A quick question. When looking at issues.asterisk.org, It allows issues/bugs to be filtered by Asterisk Version. The 1.8.x options for the filter are: 1.8.2.3 1.8.2.4 1.8.3.2 1.8.4-rc2 Do you guys know whether bugs from the older version should still show up as issues in the newer versions assuming that they weren't patched with the newer version release? In other words, if I look at the issues/bugs for 1.8.4-rc2, can I feel confident that any bugs from previous releases that are not explicitly listed for 1.8.4-rc2 have been patched somewhere between that version and 1.8.4-rc2? Or should I examine every single issue for every 1.8.x version look at the notes on each bug? Or if a bug's status is set to Closed, can I assume that the latest release (1.8.4-rc2) should not exhibit the bug? I appreciate your input. I just want to make sure that I take the correct approach to this and don't wind up with a system with bugs that I wasn't expecting. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . -- _ -- 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] checking dahdi channels
On Fri, Mar 25, 2011 at 11:36 AM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 25 Mar 2011, Nathan Pryor wrote: Is there a command I could use directly in the dialplan or with the manager interface to get the number of used channels? Check out the GROUP() and GROUP_COUNT() functions. That's what I needed. Thanks Steve! -- _ -- 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] Why shouldn't I use 1.8?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Friday, March 25, 2011 12:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why shouldn't I use 1.8? A quick question. When looking at issues.asterisk.org, It allows issues/bugs to be filtered by Asterisk Version. The 1.8.x options for the filter are: 1.8.2.3 1.8.2.4 1.8.3.2 1.8.4-rc2 Do you guys know whether bugs from the older version should still show up as issues in the newer versions assuming that they weren't patched with the newer version release? In other words, if I look at the issues/bugs for 1.8.4-rc2, can I feel confident that any bugs from previous releases that are not explicitly listed for 1.8.4-rc2 have been patched somewhere between that version and 1.8.4-rc2? Or should I examine every single issue for every 1.8.x version look at the notes on each bug? Or if a bug's status is set to Closed, can I assume that the latest release (1.8.4-rc2) should not exhibit the bug? I appreciate your input. I just want to make sure that I take the correct approach to this and don't wind up with a system with bugs that I wasn't expecting. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . Personally I would check each one as there are #1 bugs that live from version to version and #2 (and this doesn't happen too often, but it does) bugs introduced by other bug fixes. -- _ -- 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] SIP Invite and Asterisk API/Variable
On 24/03/11 05:49, Olivier CALVANO wrote: The To, To:sip:081169x...@91.121.xxx.xxx;user=phone, can i get it into a variable for sent it at a API ? You want the sip_header function: http://www.voip-info.org/wiki/view/Asterisk+func+sip_header cheers, Paul. -- _ -- 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
[asterisk-users] 3com 3102
Has anyone had any luck getting this phone up and running on an asterisk server, most noticeably a Trixbox installation? -- _ -- 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
[asterisk-users] reload command not availeble asterisk 1.8.x
Hey Guys! I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has reload command but other doesn't ? satish-desktop*CLI core show version Asterisk 1.8.3.2 built by root @ satish-desktop on a x86_64 running Linux on 2011-03-25 16:10:39 UTC satish-desktop*CLI re tabtab realtime reload shirley*CLI core show version Asterisk 1.8.3.2 built by root @ shirley on a x86_64 running Linux on 2011-03-22 18:38:19 UTC shirley*CLI re tabtab destroy load mysqlstoreupdate update2 -- _ -- 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] reload command not availeble asterisk 1.8.x
On 11-03-25 02:49 PM, satish patel wrote: I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has reload command but other doesn't ? *CLI module reload 'reload' is no longer a valid command. I'm guess one box has cli_aliases.conf, while the other does not. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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] reload command not availeble asterisk 1.8.x
Both servers files are identical.. root@satish-desktop:~# cat /etc/asterisk/cli_aliases.conf | grep reload reload=module reload ; Alias for making voicemail reload actually do module reload app_voicemail.so ;voicemail reload=module reload app_voicemail.so ; This will make the CLI command mr behave as though it is module reload. ;mr=module reload extensions reload=dialplan reload root@shirley:/# cat /etc/asterisk/cli_aliases.conf | grep reload reload=module reload ; Alias for making voicemail reload actually do module reload app_voicemail.so ;voicemail reload=module reload app_voicemail.so ; This will make the CLI command mr behave as though it is module reload. ;mr=module reload extensions reload=dialplan reload Date: Fri, 25 Mar 2011 14:57:14 -0400 From: pabelan...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] reload command not availeble asterisk 1.8.x On 11-03-25 02:49 PM, satish patel wrote: I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has reload command but other doesn't ? *CLI module reload 'reload' is no longer a valid command. I'm guess one box has cli_aliases.conf, while the other does not. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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
[asterisk-users] Asterisk with FXO card only, no network
All- My apologies in advance if this is an obvious question and I've missed it on Asterisk FAQs and how-to's... Can Asterisk operate with just an FXO card? By that I mean, no network connection (none, no local network). I want to build some type of user interface to go off-hook, route FXO port voice lines to a headset (or speaker and microphone), perform an automated conversation (e.g. playout from sound card), and go back on-hook. The reason for this is automated test -- I'm trying to find a flexible, programmable way to emulate two (2) analog handsets. The test software would make a series of calls between them (different lines/numbers connected to the 2 FXO ports). (As a side note, the test software also interfaces with other equipment (RS-232 or USB), and the whole sequence will eventually be automated and may take several hours to complete a regression test.) Again, I apologize for the dumbness of this question. We use Asterisk as a B2BUA all the time but this particular question came up recently. Thanks. -Jeff -- _ -- 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
[asterisk-users] Back-to-back asterisk PRI issue
Following is my scenario to connect back to back PRI of two asterisk server.
PRI cards are Sangoma A102D
[Asterisk1][PRI]-Cross Cable-[Asterisk2]
Asterisk1
; Span 1 (MASTER)
switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0,24
echocancel = yes
signalling = pri_net
channel = 1-23
Asterisk2
; Span 1
switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0,24
echocancel = yes
signalling = pri_cpe
channel = 1-23
Following is my extensions.conf stuff on both machine (extension number could
be change)
[from-pstn]
exten = s,1,Answer()
same = n,Playback(hello-world)
same = n,Hangup()
[from-sip]
exten = _7XXX,1,Answer()
same = n,Dial(SIP/${EXTEN})
same = n,Hangup()
exten = 7527,1,Dial(DAHDI/G0/7527)
But i am getting following error when i am calling from A to B
satish-desktop*CLI
[Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
[Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor: Cannot connect
[Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument
[Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument
--
_
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Re: [asterisk-users] reload command not availeble asterisk 1.8.x
On 11-03-25 03:13 PM, satish patel wrote: Both servers files are identical.. *CLI module show like cli -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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] reload command not availeble asterisk 1.8.x
satish-desktop*CLI module show like cli Module Description Use Count res_clioriginate.soCall origination and redirection from th 0 res_clialiases.so CLI Aliases 0 2 modules loaded shirley*CLI module show like cli Module Description Use Count res_clioriginate.soCall origination and redirection from th 0 1 modules loaded Date: Fri, 25 Mar 2011 15:45:13 -0400 From: pabelan...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] reload command not availeble asterisk 1.8.x On 11-03-25 03:13 PM, satish patel wrote: Both servers files are identical.. *CLI module show like cli -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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
Re: [asterisk-users] [SOLVED] reload command not availeble asterisk 1.8.x
No kidding.. found this line second server. Thanks!! root@shirley:/# cat /etc/asterisk/modules.conf | grep res_clialiases.so noload = res_clialiases.so From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 19:53:58 + Subject: Re: [asterisk-users] reload command not availeble asterisk 1.8.x satish-desktop*CLI module show like cli Module Description Use Count res_clioriginate.soCall origination and redirection from th 0 res_clialiases.so CLI Aliases 0 2 modules loaded shirley*CLI module show like cli Module Description Use Count res_clioriginate.soCall origination and redirection from th 0 1 modules loaded Date: Fri, 25 Mar 2011 15:45:13 -0400 From: pabelan...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] reload command not availeble asterisk 1.8.x On 11-03-25 03:13 PM, satish patel wrote: Both servers files are identical.. *CLI module show like cli -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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
Re: [asterisk-users] What is the most stable version of asterisk?
1.4 is the new flavor for my new deployments, but I definitely have more (way more, like 1:8) 1.2 systems in production. On Fri, Mar 25, 2011 at 10:32 AM, Douglas Mortensen d...@impalanetworks.com wrote: Do you have the same ratio of deployments using 1.4 as you do with 1.2? What about 1.6 or 1.8? I simply question how accurate a comparison can be made when one is comparing 50 1.2 deployments to 5 1.4 deployments? I'm sure it says something, and I do appreciate the feedback. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . -Original Message- From: C F [mailto:shma...@gmail.com] Sent: Thursday, March 24, 2011 8:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the most stable version of asterisk? I use mainly 1.2 with great success, mostly restarts are due to power outages. I recently started to upgrade to 1.4, so far so good. Too early to say, the longest running 1.4 is only 234 days. While I have had 900+ days with 1.2 -- _ -- 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
Re: [asterisk-users] Back-to-back asterisk PRI issue
Did you check so see if the pri is up?
Also, make sure wanpipe dahdi is setup correctly.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Friday, March 25, 2011 3:41 PM
To: asterisk-users
Subject: [asterisk-users] Back-to-back asterisk PRI issue
Following is my scenario to connect back to back PRI of two asterisk server.
PRI cards are Sangoma A102D
[Asterisk1][PRI]-Cross Cable-[Asterisk2]
Asterisk1
; Span 1 (MASTER)
switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0,24
echocancel = yes
signalling = pri_net
channel = 1-23
Asterisk2
; Span 1
switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0,24
echocancel = yes
signalling = pri_cpe
channel = 1-23
Following is my extensions.conf stuff on both machine (extension number
could be change)
[from-pstn]
exten = s,1,Answer()
same = n,Playback(hello-world)
same = n,Hangup()
[from-sip]
exten = _7XXX,1,Answer()
same = n,Dial(SIP/${EXTEN})
same = n,Hangup()
exten = 7527,1,Dial(DAHDI/G0/7527)
But i am getting following error when i am calling from A to B
satish-desktop*CLI
[Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
[Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor: Cannot connect
[Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument
[Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Asterisk 1.6.2.10 CDR custom added field
On Friday 25 March 2011 04:36:28 Jonas Kellens wrote: On 03/25/2011 08:19 AM, Tilghman Lesher wrote: On Thursday 24 March 2011 04:50:48 Jonas Kellens wrote: On 03/24/2011 10:45 AM, Rizwan Hisham wrote: You have to use adaptive cdr for this functionality. In 1.8 the conf file for adaptive cdr is cdr_adaptive_odbc.conf. The sample conf file should tell you everything. If you are using some other cdr engine then you will have to jump into the code of asterisk to make it log the item you want, which includes creating an extra variable in the cdr data struction, creating a function to set/get its value from dialplan, and then changing the sql command to include the extra variable for insertion into DB. I thought it was possible in asterisk 1.6.2 to add extra mysql-fields ?? In asterisk 1.4 you just have one 'userfield', but in 1.6.2 it is possible to add custom fields... I just don't know how. This is what the wiki (http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql) tells : /Module now permits arbitrary columns to be created and populated, just like cdr_adaptive_odbc, simply by adding the column to the table and defining the corresponding CDR() variable/ Where is the information on this ? Same as always, in the configs/ directory of addons 1.6.2. The sample configuration file contains common examples of the added functionality. Also, there's a note on it in UPGRADE.txt, in the root directory of addons 1.6.2. If you have any further questions, you're welcome to ask this list. alias start = calldate alias callerid = clid ;alias uniqueid = uniqueid But this is not explained... Alias allows you to rename a standard named column to another column name. I agree that the commented items are confusing. However, both of the uncommented ones are common renames of the standard columns. So please can you confirm how I think it should work : In my dialplan I have : /exten = 600,n,Set(CDR(mycolumn)=myvalue)/ So I should add the following to cdr_mysql.conf : /[columns] static mycolumn = mycolumn/ No, what this will do is add the static definition of the literal value mycolumn to the mycolumn field. What you actually want is to add the field to your table (ALTER TABLE ... ADD COLUMN ...) and add it to your extensions.conf (and reload). That's it. There is literally nothing you have to change in the cdr_mysql.conf file to add an extra column. There is also literally nothing you have to change in the cdr_mysql.conf file to _delete_ a standard column. Just have the column not appear in the backend table (ALTER TABLE ... DROP COLUMN ...) and reload Asterisk. The static definition is for implicit values only. The alias column is just for renaming standard columns. -- Tilghman -- _ -- 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] Back-to-back asterisk PRI issue
satish patel wrote: group = 0,24 Granted, I'm still running 1.4.x, but I don't believe the above is valid. My guess is it should be: group = 0 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Back-to-back asterisk PRI issue
On Friday 25 March 2011 15:11:49 Doug Lytle wrote:
satish patel wrote:
group = 0,24
Granted, I'm still running 1.4.x, but I don't believe the above is
valid.
My guess is it should be:
group = 0
No, that's valid. You can have any of groups 0-63 set on a single
group of channels. They are for group selection of channels, as in
Dial(DAHDI/g0/${EXTEN:1})
--
Tilghman
--
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Re: [asterisk-users] Back-to-back asterisk PRI issue
Asterisk1
satish-desktop*CLI dahdi show status
Description Alarms IRQbpviol CRC4 Fra Codi
Options LBO
wanpipe1 card 0 OK 0 0 0 ESF B8ZS
0 db (CSU)/0-133 feet (DSX-1)
wanpipe2 card 1 UNCONFI 0 0 0 CAS Unk
0 db (CSU)/0-133 feet (DSX-1)
satish-desktop*CLI
Asterisk2
shirley*CLI dahdi show status
Description Alarms IRQbpviol CRC4 Fra Codi
Options LBO
wanpipe1 card 0 OK 0 0 0 ESF B8ZS
0 db (CSU)/0-133 feet (DSX-1)
wanpipe2 card 1 RED 0 0 0 CAS Unk
0 db (CSU)/0-133 feet (DSX-1)
shirley*CLI
From: will...@stillwellsoft.com
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 16:04:12 -0400
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
Did you check so see if the pri is up? Also, make sure wanpipe dahdi is setup
correctly. From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Friday, March 25, 2011 3:41 PM
To: asterisk-users
Subject: [asterisk-users] Back-to-back asterisk PRI issue Following is my
scenario to connect back to back PRI of two asterisk server. PRI cards are
Sangoma A102D
[Asterisk1][PRI]-Cross Cable-[Asterisk2]
Asterisk1
; Span 1 (MASTER)
switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0,24
echocancel = yes
signalling = pri_net
channel = 1-23
Asterisk2
; Span 1
switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0,24
echocancel = yes
signalling = pri_cpe
channel = 1-23
Following is my extensions.conf stuff on both machine (extension number could
be change)
[from-pstn]
exten = s,1,Answer()
same = n,Playback(hello-world)
same = n,Hangup()
[from-sip]
exten = _7XXX,1,Answer()
same = n,Dial(SIP/${EXTEN})
same = n,Hangup()
exten = 7527,1,Dial(DAHDI/G0/7527)
But i am getting following error when i am calling from A to B
satish-desktop*CLI
[Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
[Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor: Cannot connect
[Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument
[Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument
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Re: [asterisk-users] Back-to-back asterisk PRI issue
Thanks Doug, I tried that also but result is same. Date: Fri, 25 Mar 2011 16:11:49 -0400 From: supp...@drdos.info To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue satish patel wrote: group = 0,24 Granted, I'm still running 1.4.x, but I don't believe the above is valid. My guess is it should be: group = 0 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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
Re: [asterisk-users] Back-to-back asterisk PRI issue
On Friday 25 March 2011 14:40:40 satish patel wrote: Following is my scenario to connect back to back PRI of two asterisk server. PRI cards are Sangoma A102D [Asterisk1][PRI]-Cross Cable-[Asterisk2] Asterisk1 ; Span 1 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_net channel = 1-23 Asterisk2 ; Span 1 switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_cpe channel = 1-23 Here's one confusing part. You're saying that calls that come from the master to the slave end up in context from-pstn (on the slave), but calls from the slave to the master ALSO end up in from-pstn (on the master). Seems like one of them should be from-internal or the like. I'm sure some of your problem emanate from these settings. satish-desktop*CLI [Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) Check the other side for error messages. [Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor: Cannot connect [Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument [Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument This problem is due to a misconfiguration. Asterisk cannot handle the local network being addressed as the 0.0.0.0 network. You need to use the full local address. -- Tilghman -- _ -- 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] Back-to-back asterisk PRI issue
One more thing i would like to tell you i have following wanpipe configuration at both side @Asterisk1 root@satish-desktop:~# cat /etc/wanpipe/wanpipe1.conf | grep -i clock TE_CLOCK= MASTER TE_REF_CLOCK= 0 @Asterisk2 root@shirley:/# cat /etc/wanpipe/wanpipe2.conf | grep -i clock TE_CLOCK= NORMAL TE_REF_CLOCK= 0 From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 20:25:31 + Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue Thanks Doug, I tried that also but result is same. Date: Fri, 25 Mar 2011 16:11:49 -0400 From: supp...@drdos.info To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue satish patel wrote: group = 0,24 Granted, I'm still running 1.4.x, but I don't believe the above is valid. My guess is it should be: group = 0 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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
Re: [asterisk-users] Back-to-back asterisk PRI issue
Okay! i have changed context at master side ; Span 1: WPT1/0 wanpipe1 card 0 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-internal group = 1,24 echocancel = yes signalling = pri_net channel = 1-23 Same error nothing change.. satish-desktop*CLI core set verbose 10 Verbosity was 0 and is now 10 satish-desktop*CLI core set debug 999 Core debug was 0 and is now 999 == Using SIP RTP CoS mark 5 -- Executing [7527@from-sip:1] Dial(SIP/7623-, DAHDI/g1/527) in new stack [Mar 25 16:39:47] WARNING[5336]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/7623-' status is 'CONGESTION' From: tilgh...@meg.abyt.es To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 15:35:21 -0500 Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue On Friday 25 March 2011 14:40:40 satish patel wrote: Following is my scenario to connect back to back PRI of two asterisk server. PRI cards are Sangoma A102D [Asterisk1][PRI]-Cross Cable-[Asterisk2] Asterisk1 ; Span 1 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_net channel = 1-23 Asterisk2 ; Span 1 switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_cpe channel = 1-23 Here's one confusing part. You're saying that calls that come from the master to the slave end up in context from-pstn (on the slave), but calls from the slave to the master ALSO end up in from-pstn (on the master). Seems like one of them should be from-internal or the like. I'm sure some of your problem emanate from these settings. satish-desktop*CLI [Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) Check the other side for error messages. [Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor: Cannot connect [Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument [Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument This problem is due to a misconfiguration. Asterisk cannot handle the local network being addressed as the 0.0.0.0 network. You need to use the full local address. -- Tilghman -- _ -- 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
Re: [asterisk-users] Back-to-back asterisk PRI issue
Only Difference is one side card is ECHO Cancellation supported and other is non-ECHO cancellation. Is there any issue ? @Asterisk1 Sangoma A102 (non-ECHW) @Asterisk2 Sangoma A102D (ECHW) -Satish From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 20:41:09 + Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue Okay! i have changed context at master side ; Span 1: WPT1/0 wanpipe1 card 0 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-internal group = 1,24 echocancel = yes signalling = pri_net channel = 1-23 Same error nothing change.. satish-desktop*CLI core set verbose 10 Verbosity was 0 and is now 10 satish-desktop*CLI core set debug 999 Core debug was 0 and is now 999 == Using SIP RTP CoS mark 5 -- Executing [7527@from-sip:1] Dial(SIP/7623-, DAHDI/g1/527) in new stack [Mar 25 16:39:47] WARNING[5336]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/7623-' status is 'CONGESTION' From: tilgh...@meg.abyt.es To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 15:35:21 -0500 Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue On Friday 25 March 2011 14:40:40 satish patel wrote: Following is my scenario to connect back to back PRI of two asterisk server. PRI cards are Sangoma A102D [Asterisk1][PRI]-Cross Cable-[Asterisk2] Asterisk1 ; Span 1 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_net channel = 1-23 Asterisk2 ; Span 1 switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_cpe channel = 1-23 Here's one confusing part. You're saying that calls that come from the master to the slave end up in context from-pstn (on the slave), but calls from the slave to the master ALSO end up in from-pstn (on the master). Seems like one of them should be from-internal or the like. I'm sure some of your problem emanate from these settings. satish-desktop*CLI [Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) Check the other side for error messages. [Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor: Cannot connect [Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument [Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument This problem is due to a misconfiguration. Asterisk cannot handle the local network being addressed as the 0.0.0.0 network. You need to use the full local address. -- Tilghman -- _ -- 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
Re: [asterisk-users] Back-to-back asterisk PRI issue
sometime i am getting following error also. what is this means? [Mar 25 17:11:52] WARNING[5961]: sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway! From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 21:04:45 + Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue Only Difference is one side card is ECHO Cancellation supported and other is non-ECHO cancellation. Is there any issue ? @Asterisk1 Sangoma A102 (non-ECHW) @Asterisk2 Sangoma A102D (ECHW) -Satish From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 20:41:09 + Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue Okay! i have changed context at master side ; Span 1: WPT1/0 wanpipe1 card 0 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-internal group = 1,24 echocancel = yes signalling = pri_net channel = 1-23 Same error nothing change.. satish-desktop*CLI core set verbose 10 Verbosity was 0 and is now 10 satish-desktop*CLI core set debug 999 Core debug was 0 and is now 999 == Using SIP RTP CoS mark 5 -- Executing [7527@from-sip:1] Dial(SIP/7623-, DAHDI/g1/527) in new stack [Mar 25 16:39:47] WARNING[5336]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/7623-' status is 'CONGESTION' From: tilgh...@meg.abyt.es To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 15:35:21 -0500 Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue On Friday 25 March 2011 14:40:40 satish patel wrote: Following is my scenario to connect back to back PRI of two asterisk server. PRI cards are Sangoma A102D [Asterisk1][PRI]-Cross Cable-[Asterisk2] Asterisk1 ; Span 1 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_net channel = 1-23 Asterisk2 ; Span 1 switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_cpe channel = 1-23 Here's one confusing part. You're saying that calls that come from the master to the slave end up in context from-pstn (on the slave), but calls from the slave to the master ALSO end up in from-pstn (on the master). Seems like one of them should be from-internal or the like. I'm sure some of your problem emanate from these settings. satish-desktop*CLI [Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) Check the other side for error messages. [Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor: Cannot connect [Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument [Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument This problem is due to a misconfiguration. Asterisk cannot handle the local network being addressed as the 0.0.0.0 network. You need to use the full local address. -- Tilghman -- _ -- 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:
Re: [asterisk-users] Back-to-back asterisk PRI issue
I just start Pri set debug on span 1 and its showing D-channel is down satish-desktop*CLI pri show span Usage: pri show span span Displays PRI Information on a given PRI span satish-desktop*CLI pri show span 1 Primary D-channel: 24 Status: Down, Active Switchtype: Q.SIG switch Type: Network Overlap Dial: 0 Logical Channel Mapping: 0 Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T202: 1 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000 T313: 4000 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 Overlap Recv: No satish-desktop*CLI pri set debug on span 1 1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) 1 Changing from state 5(Awaiting establishment) to 4(TEI assigned) 1 TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) 1 TEI=0 Sending SABME 1 Changing from state 4(TEI assigned) to 5(Awaiting establishment) Span: 1 Processing event: PRI_EVENT_DCHAN_DOWN 1 TEI=0 Sending SABME 1 TEI=0 Sending SABME 1 TEI=0 Sending SABME 1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) 1 Changing from state 5(Awaiting establishment) to 4(TEI assigned) 1 TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) 1 TEI=0 Sending SABME 1 Changing from state 4(TEI assigned) to 5(Awaiting establishment) Span: 1 Processing event: PRI_EVENT_DCHAN_DOWN 1 TEI=0 Sending SABME 1 TEI=0 Sending SABME 1 TEI=0 Sending SABME 1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) 1 Changing from state 5(Awaiting establishment) to 4(TEI assigned) 1 TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) 1 TEI=0 Sending SABME 1 Changing from state 4(TEI assigned) to 5(Awaiting establishment) Span: 1 Processing event: PRI_EVENT_DCHAN_DOWN 1 TEI=0 Sending SABME 1 TEI=0 Sending SABME 1 TEI=0 Sending SABME 1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) 1 Changing from state 5(Awaiting establishment) to 4(TEI assigned) 1 TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) 1 TEI=0 Sending SABME 1 Changing from state 4(TEI assigned) to 5(Awaiting establishment) Span: 1 Processing event: PRI_EVENT_DCHAN_DOWN 1 TEI=0 Sending SABME 1 TEI=0 Sending SABME 1 TEI=0 Sending SABME 1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) 1 Changing from state 5(Awaiting establishment) to 4(TEI assigned) 1 TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) 1 TEI=0 Sending SABME 1 Changing from state 4(TEI assigned) to 5(Awaiting establishment) Span: 1 Processing event: PRI_EVENT_DCHAN_DOWN 1 TEI=0 Sending SABME 1 TEI=0 Sending SABME From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 21:13:34 + Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue sometime i am getting following error also. what is this means? [Mar 25 17:11:52] WARNING[5961]: sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway! From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 21:04:45 + Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue Only Difference is one side card is ECHO Cancellation supported and other is non-ECHO cancellation. Is there any issue ? @Asterisk1 Sangoma A102 (non-ECHW) @Asterisk2 Sangoma A102D (ECHW) -Satish From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 20:41:09 + Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue Okay! i have changed context at master side ; Span 1: WPT1/0 wanpipe1 card 0 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-internal group = 1,24 echocancel = yes signalling = pri_net channel = 1-23 Same error nothing change.. satish-desktop*CLI core set verbose 10 Verbosity was 0 and is now 10 satish-desktop*CLI core set debug 999 Core debug was 0 and is now 999 == Using SIP RTP CoS mark 5 -- Executing [7527@from-sip:1] Dial(SIP/7623-, DAHDI/g1/527) in new stack [Mar 25 16:39:47] WARNING[5336]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/7623-' status is 'CONGESTION' From: tilgh...@meg.abyt.es To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 15:35:21 -0500 Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue On Friday 25 March 2011 14:40:40 satish patel wrote: Following is my scenario to connect back to back PRI of two asterisk server. PRI cards are Sangoma A102D [Asterisk1][PRI]-Cross Cable-[Asterisk2] Asterisk1 ; Span 1 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_net channel
Re: [asterisk-users] Back-to-back asterisk PRI issue
On Friday 25 March 2011 16:23:27 satish patel wrote: I just start Pri set debug on span 1 and its showing D-channel is down How do you have the underlying T1 signalling set up in /etc/dahdi/system.conf (on both ends)? -- Tilghman -- _ -- 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
[asterisk-users] Removing Polycom Transfer Softkey
Sorry for the crosspost. This was supposed to be on -users I know some of you are polycom gurus... Anyone know how to remove transfer from a polycom 33x phone? We've set allowtransfer=no, but we would like to remove a polycom soft key as well. -- _ -- 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
[asterisk-users] White papers or success cases to convince a customer?
Can anyone recommend some White Papers or Success Cases that we can use to ease the mind of a customer that has not heard much about Asterisk? All they know is Avaya at this point. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- 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] Back-to-back asterisk PRI issue
Both server has same content in system.conf file. satish@shirley:~$ cat /etc/dahdi/system.conf # Global data loadzone= us defaultzone = us span = 1,1,0,esf,b8zs bchan = 1-23 dchan=24 echocanceller = mg2,1-23 From: tilgh...@meg.abyt.es To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 17:23:28 -0500 Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue On Friday 25 March 2011 16:23:27 satish patel wrote: I just start Pri set debug on span 1 and its showing D-channel is down How do you have the underlying T1 signalling set up in /etc/dahdi/system.conf (on both ends)? -- Tilghman -- _ -- 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
Re: [asterisk-users] White papers or success cases to convince a customer?
On Fri, Mar 25, 2011 at 6:05 PM, Carlos Chavez cur...@telecomabmex.comwrote: Can anyone recommend some White Papers or Success Cases that we can use to ease the mind of a customer that has not heard much about Asterisk? All they know is Avaya at this point. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 Hopefully someone can point you to some papers, but if you end up just needing someone to write up a testimonial about how Asterisk has been a successful part of several (20+) projects, contact me offlist and I'll write one up. I've used Asterisk in projects ranging from very small business PBX's all the way up to large VOIP telephone service providers. -- _ -- 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] Back-to-back asterisk PRI issue
Check out this https://issues.asterisk.org/view.php?id=17270 From: tilgh...@meg.abyt.es To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 17:23:28 -0500 Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue On Friday 25 March 2011 16:23:27 satish patel wrote: I just start Pri set debug on span 1 and its showing D-channel is down How do you have the underlying T1 signalling set up in /etc/dahdi/system.conf (on both ends)? -- Tilghman -- _ -- 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
Re: [asterisk-users] White papers or success cases to convince a customer?
On Fri, Mar 25, 2011 at 7:05 PM, Carlos Chavez cur...@telecomabmex.com wrote: Can anyone recommend some White Papers or Success Cases that we can use to ease the mind of a customer that has not heard much about Asterisk? All they know is Avaya at this point. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 There is a small list here http://www.digium.com/en/company/casestudies/ I would suggest you watch the Keynote speech by Kevin at the last Astricon... I think he mentions some numbers... -- ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- 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
[asterisk-users] pbx.c: We were unable to say the number
Hello, Occasionally, I get the following warning in my asterisk log, pbx.c: We were unable to say the number [n], is it too large? n is two or one digit number, which doesn't look like large to me! Could anybody please tell more about this warning, like in what scenario I may have this warning. Thanks, Mohammad -- _ -- 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
