[Asterisk-Users] Re: I've broken voicemail
spam at dynsysgroup.com
spam at dynsysgroup.com
Tue May 23 08:20:28 MST 2006
May 22 23:07:50 WARNING[3119]: Unexpected freqency 44100
I can't believe i didn't see that!
i spent ages staring at those damn logs...
i'm sure that will fix it.
thanks
r
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> Today's Topics:
>
> 1. Initial second lost on SIP phones (Pieter Claassen)
> 2. How to detect call forwarding to voicemail (Nitin Gupta)
> 3. I've broken voicemail (Robbie Hughes)
> 4. Re: Re: I get MOH when the caller hangs up (Steve Totaro)
> 5. Re: How to detect call forwarding to voicemail
> (Eric "ManxPower" Wieling)
> 6. Re: Office to Office via IAX2 problems (asterisk at txpe.net)
> 7. Re: I've broken voicemail (Patrick)
> 8. FAX and Asterisk (Rene Nelson)
> 9. Re: I've broken voicemail (Mark Phillips)
> 10. Re: FAX and Asterisk (Lee Howard)
> 11. Re: How to detect call forwarding to voicemail (Leo Ann Boon)
> 12. Re: Office to Office via IAX2 problems (asterisk at txpe.net)
> 13. Timeframe for QueueStatus values (mbodbg at gmx.net)
> 14. Re: Timeframe for QueueStatus values (BJ Weschke)
> 15. CallerID (Greg Oliver)
> 16. Re: Help Avaya 4606 (Tom Lynn)
> 17. US telco lingo (Eric Bishop)
> 18. RE: US telco lingo (Kerry Garrison)
> 19. Faxing - machines stop talking, line stays up (Warrick Zedi)
> 20. Re: FAX and Asterisk (Alejandro Vargas)
> 21. TDM400P , "ztcfg ?vv error ", "Does it have to do with my PC
> hardware ?" (John Joseph)
> 22. [Fwd: [Asterisk-Users] Faxing - machines stop talking, line
> stays up] (Warrick Zedi)
> 23. SIP session number (Giordano Grandis)
> 24. Logger rotate & master.csv (Asterisk)
> 25. Free/Open pci telco card (Kai Ober)
> 26. Re: voicemail access on the Thomson ST2030 ?
> (Louis-David Mitterrand)
> 27. A call from a call file always does a redial? (Remco Barende)
> 28. Re: Deadlocks in 1.2.7.1 (Philipp Ott)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 22 May 2006 23:27:36 +0200
> From: Pieter Claassen <pieter at claassen.co.uk>
> Subject: [Asterisk-Users] Initial second lost on SIP phones
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <200605222327.36237.pieter at claassen.co.uk>
> Content-Type: text/plain; charset="us-ascii"
>
> I find that when asterisk answers the phone, the initial second or so is
> lost.
> I can imagine that echotraining can do this, but this is between SIP
> phones
> and I don't think there is any echotraining enabled?
>
> BTW. Asterisk is definitely playing sounds that first second (The CLI
> would
> indicate that it would play a beep but I just won't hear it).
>
> Any comments appreciated.
> Pieter
>
>
> ------------------------------
>
> Message: 2
> Date: Mon, 22 May 2006 14:31:17 -0700
> From: "Nitin Gupta" <niting at gmail.com>
> Subject: [Asterisk-Users] How to detect call forwarding to voicemail
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <e37899da0605221431s635a47f4j970451373435b836 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
> Is there anyway in Asterisk to know that outgoing call has been forwarded
> to voicemail by the callee system?
>
> Some of my users don't want to connect the call if its forwarded to callee
> voicemail, so I am wondering if theres anyway to identify this in Asterisk
> and drop the call.
>
> Thanks
> Nitin
> -------------- next part --------------
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>
> ------------------------------
>
> Message: 3
> Date: Mon, 22 May 2006 23:11:15 +0100
> From: Robbie Hughes <spam at dynsysgroup.com>
> Subject: [Asterisk-Users] I've broken voicemail
> To: "asterisk-users at lists.digium.com"
> <asterisk-users at lists.digium.com>
> Message-ID: <C097F593.15BA%spam at dynsysgroup.com>
> Content-Type: text/plain; charset="US-ASCII"
>
> I went to put in the new sound files over the weekend, but forgot to
> backup
> the custom folder and lost my custom digital receptionist files.
> I then had to copy the old files back from a duplicate machine.
>
> The problem is now though that voicemail just hangs up when I dial it.
>
> Other apps work - *70 for example gives me 'call waiting...activated' so I
> know it's accessing the files correctly, and the gsm files themselves are
> all chown asterisk, chgrp asterisk chmod 644 * etc so they can be read and
> there shouldn't be a permissions issue...but voicemail is not working.
>
> In asterisk I get the following:
>
>
> asterisk*CLI> set verbose 999
> Verbosity was 0 and is now 999
> -- Executing Answer("SIP/1607-a359", "") in new stack
> -- Executing Wait("SIP/1607-a359", "1") in new stack
> -- Executing VoiceMailMain("SIP/1607-a359", "default") in new stack
> == Spawn extension (from-internal, *98, 3) exited non-zero on
> 'SIP/1607-a359'
> -- Executing Macro("SIP/1607-a359", "hangupcall") in new stack
> -- Executing ResetCDR("SIP/1607-a359", "w") in new stack
> -- Executing NoCDR("SIP/1607-a359", "") in new stack
> -- Executing Wait("SIP/1607-a359", "5") in new stack
> -- Executing Hangup("SIP/1607-a359", "") in new stack
> == Spawn extension (macro-hangupcall, s, 4) exited non-zero on
> 'SIP/1607-a359' in macro 'hangupcall'
> == Spawn extension (from-internal, h, 1) exited non-zero on
> 'SIP/1607-a359'
>
>
> And full log gives me
>
> May 22 23:07:49 DEBUG[3119]: Setting NAT on RTP to 0
> May 22 23:07:49 DEBUG[3119]: Stopping retransmission on
> '5F145956-E9DF-11DA-84DB-0016CB90EF7F at 192.168.4.144' of Response 3706:
> Found
> May 22 23:07:49 DEBUG[3119]: Setting NAT on RTP to 0
> May 22 23:07:49 DEBUG[3119]: Check for res for 1607
> May 22 23:07:49 DEBUG[3119]: Call from user '1607' is 1 out of 0
> May 22 23:07:49 DEBUG[3119]: build_route: Contact hop:
> <sip:1607 at 192.168.4.144:5060>
> May 22 23:07:49 VERBOSE[3119]: -- Executing Answer("SIP/1607-bd04",
> "")
> in new stack
> May 22 23:07:49 VERBOSE[3119]: -- Executing Wait("SIP/1607-bd04", "1")
> in new stack
> May 22 23:07:49 DEBUG[3119]: Stopping retransmission on
> '5F145956-E9DF-11DA-84DB-0016CB90EF7F at 192.168.4.144' of Response 3707:
> Found
> May 22 23:07:50 VERBOSE[3119]: -- Executing
> VoiceMailMain("SIP/1607-bd04", "default") in new stack
> May 22 23:07:50 WARNING[3119]: Unexpected freqency 44100
> May 22 23:07:50 WARNING[3119]: Unable to open fd on
> /var/lib/asterisk/sounds/vm-login.wav
> May 22 23:07:50 WARNING[3119]: Unable to open vm-login (format ulaw): No
> such file or directory
> May 22 23:07:50 WARNING[3119]: Couldn't stream login file
> May 22 23:07:50 VERBOSE[3119]: == Spawn extension (from-internal, *98,
> 3)
> exited non-zero on 'SIP/1607-bd04'
> May 22 23:07:50 VERBOSE[3119]: -- Executing Macro("SIP/1607-bd04",
> "hangupcall") in new stack
> May 22 23:07:50 VERBOSE[3119]: -- Executing ResetCDR("SIP/1607-bd04",
> "w") in new stack
> May 22 23:07:50 DEBUG[3119]: cdr_mysql: inserting a CDR record.
> May 22 23:07:50 DEBUG[3119]: cdr_mysql: SQL command as follows: INSERT
> INTO
> cdr
> (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration
> ,billsec,disposition,amaflags,accountcode) VALUES ('2006-05-22
> 23:07:49','\"GPPlus\" <1607>','1607','*98','from-internal',
> 'SIP/1607-bd04','','ResetCDR','w',1,1,'ANSWERED',3,'')
> May 22 23:07:50 VERBOSE[3119]: -- Executing NoCDR("SIP/1607-bd04", "")
> in new stack
> May 22 23:07:50 WARNING[3119]: CDR on channel 'SIP/1607-bd04' not posted
> May 22 23:07:50 WARNING[3119]: CDR on channel 'SIP/1607-bd04' lacks end
> May 22 23:07:50 VERBOSE[3119]: -- Executing Wait("SIP/1607-bd04", "5")
> in new stack
> May 22 23:07:55 VERBOSE[3119]: -- Executing Hangup("SIP/1607-bd04",
> "")
> in new stack
> May 22 23:07:55 VERBOSE[3119]: == Spawn extension (macro-hangupcall, s,
> 4)
> exited non-zero on 'SIP/1607-bd04' in macro 'hangupcall'
> May 22 23:07:55 VERBOSE[3119]: == Spawn extension (from-internal, h, 1)
> exited non-zero on 'SIP/1607-bd04'
> May 22 23:07:55 DEBUG[3119]: update_user_counter(1607) - decrement inUse
> counter
> May 22 23:07:55 DEBUG[3119]: Stopping retransmission on
> '5F145956-E9DF-11DA-84DB-0016CB90EF7F at 192.168.4.144' of Request 102: Found
>
>
>
> But nothing there is particularly life-threatening...that I can see at
> least?
>
> Does anyone have any ideas of where I could start to look for this
> problem?
> I can't find any evidence that I've broken anything but for the fact that
> voicemail is broken.
>
> Any help appreciated.
> r
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Mon, 22 May 2006 18:17:19 -0400
> From: Steve Totaro <stotaro at asteriskhelpdesk.com>
> Subject: Re: [Asterisk-Users] Re: I get MOH when the caller hangs up
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4472386F.9060209 at asteriskhelpdesk.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Douglas Garstang wrote:
>> Do you have a 'g' option in your dial command? That will cause the dial
>> plan to continue executing after they hangup.... I think.
>>
>>
>>> -----Original Message-----
>>> From: Tony Mountifield [mailto:tony at softins.clara.co.uk]
>>> Sent: Monday, May 22, 2006 8:15 AM
>>> To: asterisk-users at lists.digium.com
>>> Subject: [Asterisk-Users] Re: I get MOH when the caller hangs up
>>>
>>>
>>> In article
>>> <2C2595A0F39ADC4E84623DA5CC0DB7A42A3110 at exchange.qfirst.com>,
>>> Steven Totaro <stotaro at bluehippo.com> wrote:
>>>
>>>> Exten = h,1,hangup ?
>>>>
>>> No, there's never any need to call Hangup in the h extension, because
>>> by the time h is called, the call is already hung up, by definition.
>>>
>>> Cheers
>>> Tony
>>> --
>>> Tony Mountifield
>>>
> Would it not be logical that the hangup in the h extension would hang up
> the local channel? If the local leg of the call was hungup then why MOH?
>
>
> ------------------------------
>
> Message: 5
> Date: Mon, 22 May 2006 17:21:00 -0500
> From: "Eric \"ManxPower\" Wieling" <eric at fnords.org>
> Subject: Re: [Asterisk-Users] How to detect call forwarding to
> voicemail
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4472394C.7030008 at fnords.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Nitin Gupta wrote:
>> Hi,
>> Is there anyway in Asterisk to know that outgoing call has been
>> forwarded
>> to voicemail by the callee system?
>>
>> Some of my users don't want to connect the call if its forwarded to
>> callee
>> voicemail, so I am wondering if theres anyway to identify this in
>> Asterisk
>> and drop the call.
>
> Your first question should be "Can the telco inform the calling
> equipment that the call has been forwarded to voicemail?" As far as I
> know, the answer to that is "no".
>
> --
> Now accepting new clients in Birmingham, Atlanta, Huntsville,
> Chattanooga, and Montgomery.
>
>
> ------------------------------
>
> Message: 6
> Date: Mon, 22 May 2006 16:55:12 -0500
> From: asterisk at txpe.net
> Subject: Re: [Asterisk-Users] Office to Office via IAX2 problems
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <6.2.3.4.2.20060522161126.0296c450 at mail.txpe.net>
> Content-Type: text/plain; charset="us-ascii"; format=flowed
>
> Thanks, Noah.
>
> I'm at Office A. I ssh into Office B * box using putty. While
> logged on to Office B via putty, I can ssh back into Office A * box
> by typing ssh root at officea.kicks-ass.net.
>
> Ping times can be 50-1000ms. I've tried qualify=yes, no qualify
> statement at all, qualify=1500, and qualify=2000 none helped. Each
> time I make a change, I issue the reload command from the
> CLI. Should I use a different command?
>
> Thanks,
> Doug
>
> At 03:30 PM 5/22/2006, you wrote:
>>Hi Doug -
>>
>>Just to cover all the bases. Can one machine talk to the other at
>>all? Can you ssh from one box to another (if you don't use ssh, can
>>you telnet to an open tcp port)? If not, it is surely a routing
>>issue.
>>
>>If you can connect via non-asterisk methods, you might try increasing
>>your qualify value to something higher (qualify=1500), or just remove
>>it altogether for testing. It might be that the latency is high
>>enough that the connection consistently fails to qualify. (What are
>>the ping times, BTW?)
>>
>>I'll second Eric's advice to not use a DNS name for the host, even in
>>your final setup.
>>
>>- Noah
>
>
>
>
> ------------------------------
>
> Message: 7
> Date: Tue, 23 May 2006 00:49:36 +0200
> From: Patrick <asterisk at puzzled.xs4all.nl>
> Subject: Re: [Asterisk-Users] I've broken voicemail
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1148338176.3123.2.camel at speedy.puzzled.xs4all.nl>
> Content-Type: text/plain
>
> On Mon, 2006-05-22 at 23:11 +0100, Robbie Hughes wrote:
> [snip]
>> May 22 23:07:50 WARNING[3119]: Unexpected freqency 44100
>
> Aren't Asterisk sound files supposed to use 8KHz? Did you perhaps forgot
> to transform the wav(s) you are using now to 8KHz versions?
>
> Regards,
> Patrick
>
>
>
>
>
> ------------------------------
>
> Message: 8
> Date: Mon, 22 May 2006 16:51:35 -0600
> From: "Rene Nelson" <neririn at gmail.com>
> Subject: [Asterisk-Users] FAX and Asterisk
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <60e826da0605221551h75434315u5a071533d438b3bb at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I want to accept faxes via SIP/IAX2 (yes I've read the posts that it isnt
> reccomended). My PBX is 100% Virtual with the exception of one IAX
> connection to bring in the calls from another * box, I have no phone
> hardware. I am interested in doing autodetect fax to email. I have found
> all kinds of posts to this or that, but all seem to reference Asterisk
> 1.0.xor
> 1.1.x I am running 1.2.x.
>
> Can anyone point me in the right direction to get this solution up and
> running?
>
> Thanks
> -------------- next part --------------
> An HTML attachment was scrubbed...
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>
> ------------------------------
>
> Message: 9
> Date: Mon, 22 May 2006 18:52:45 -0400
> From: Mark Phillips <g7ltt at g7ltt.com>
> Subject: Re: [Asterisk-Users] I've broken voicemail
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1148338365.17662.1.camel at earth.g7ltt.com>
> Content-Type: text/plain
>
> I know what he's done.
>
> He's installed my Alison Keenan wav files without converting them. Try
> downloading the sln files instead.
>
> BTW, G723 and G729 files going up tomorrow.
>
> Mark
>
> On Tue, 2006-05-23 at 00:49 +0200, Patrick wrote:
>> On Mon, 2006-05-22 at 23:11 +0100, Robbie Hughes wrote:
>> [snip]
>> > May 22 23:07:50 WARNING[3119]: Unexpected freqency 44100
>>
>> Aren't Asterisk sound files supposed to use 8KHz? Did you perhaps forgot
>> to transform the wav(s) you are using now to 8KHz versions?
>>
>> Regards,
>> Patrick
>>
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
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>>
>>
>
>
>
> ------------------------------
>
> Message: 10
> Date: Mon, 22 May 2006 16:05:22 -0700
> From: Lee Howard <faxguy at howardsilvan.com>
> Subject: Re: [Asterisk-Users] FAX and Asterisk
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <447243B2.805 at howardsilvan.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Rene Nelson wrote:
>
>> I want to accept faxes via SIP/IAX2 (yes I've read the posts that it
>> isnt reccomended). My PBX is 100% Virtual with the exception of one
>> IAX connection to bring in the calls from another * box, I have no
>> phone hardware. I am interested in doing autodetect fax to email. I
>> have found all kinds of posts to this or that, but all seem to
>> reference Asterisk 1.0.x or 1.1.x I am running 1.2.x.
>>
>> Can anyone point me in the right direction to get this solution up and
>> running?
>
>
> The right thing to do first would be to contact your SIP/IAX2 provider
> and find out if your connection to them will always be jitter-free (and
> this is not likely to be the case). If it will be then you can do
> faxing to their equipment. If not (and this is probably the case), then
> you will be wasting a lot of time faxing that way and you should
> probably just get a fax account with an on-line fax service.
>
> Lee.
>
>
> ------------------------------
>
> Message: 11
> Date: Tue, 23 May 2006 07:29:26 +0800
> From: Leo Ann Boon <leo at datvoiz.com>
> Subject: Re: [Asterisk-Users] How to detect call forwarding to
> voicemail
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <44724956.40009 at datvoiz.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Eric "ManxPower" Wieling wrote:
>
>>
>> Your first question should be "Can the telco inform the calling
>> equipment that the call has been forwarded to voicemail?" As far as I
>> know, the answer to that is "no".
>>
> It's possible to detect call forwarding on ISDN. The telco must enable
> it for you (for a large recurring fee when I last investigated).
> Basically, you need to compare the actual connected number vs the number
> that was dialed. I did a POC using an ISDN BRI line provided by Singtel.
> My Asterisk setup at that time was Asterisk 1.0.3 with chan-capi.
>
> Hope this helps.
>
>
>
>
>
> ------------------------------
>
> Message: 12
> Date: Mon, 22 May 2006 18:29:37 -0500
> From: asterisk at txpe.net
> Subject: Re: [Asterisk-Users] Office to Office via IAX2 problems
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <6.2.3.4.2.20060522182741.029510e0 at mail.txpe.net>
> Content-Type: text/plain; charset="us-ascii"; format=flowed
>
> Well, I changed host=ip address, qualify=2000, and rebooted computer
> and now it connects. Hopefully, this will allow Office B to stay
> connected to Office A.
>
> Thanks everyone.
>
> Doug
>
> At 03:30 PM 5/22/2006, you wrote:
>>Hi Doug -
>>
>>>Office A routinely looses connection to Office B. When typing IAX2
>>>Show Peers, it will show as Unreachable. I issue IAX2 Reload and it
>>>will work again for 1-3 days (haven't narrowed the time down yet). My
>>>theory is that the DSL at Office2 is changing IP addresses regularly
>>>and this is the cause of the problem??? This has been going on since
>>>I set up Office B (2-3 weeks). I never had to touch Office B box.
>>>Office B seemed to maintain connection, until now (see Issue 2).
>>
>>Just to cover all the bases. Can one machine talk to the other at
>>all? Can you ssh from one box to another (if you don't use ssh, can
>>you telnet to an open tcp port)? If not, it is surely a routing
>>issue.
>>
>>If you can connect via non-asterisk methods, you might try increasing
>>your qualify value to something higher (qualify=1500), or just remove
>>it altogether for testing. It might be that the latency is high
>>enough that the connection consistently fails to qualify. (What are
>>the ping times, BTW?)
>>
>>I'll second Eric's advice to not use a DNS name for the host, even in
>>your final setup.
>>
>>- Noah
>>
>>
>>On 5/22/06, Lacy Moore - Aspendora <aspendora at gmail.com> wrote:
>>>
>>>SInce you say it was working, I am assuming that both
>>> officea.kicks-ass.net
>>>and officeb.kicks-ass.net resolves to the real IP address and not an
>>>internal address, correct?
>>>
>>>Also, are you providing DNS or someone else? Is this domain registered
>>> to
>>>you? I ask that because if it is not, and you are not providing DNS, it
>>> may
>>>be resolving to another IP address. But, since you said it is the same
>>>using an IP address, this should not be the real issue.
>>>
>>>I'm not sure this would really have anything to do with it, but, if it
>>> was
>>>me, I would not have the two offices on the same subnet. I'd use
>>> 192.168.1
>>>for one and 192.168.2 for the other. It just keeps things a little
>>> simpler
>>>routing wise.
>
>
>
>
> ------------------------------
>
> Message: 13
> Date: Tue, 23 May 2006 02:25:35 +0200
> From: <mbodbg at gmx.net>
> Subject: [Asterisk-Users] Timeframe for QueueStatus values
> To: <asterisk-users at lists.digium.com>
> Message-ID: <000001c67dff$6abc0fc0$0202a8c0 at mblaptop>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello all,
>
> I've a question regarding the values "completed" and "abandoned" that are
> returned by the manager command "queuestatus". What is the timeframe for
> these values, are they counted since the last asterisk boot, or per day,
> or
> is the timeframe configurable?
>
> Thanks and Regards
>
> Markus
>
>
>
>
>
>
>
> ------------------------------
>
> Message: 14
> Date: Mon, 22 May 2006 20:57:44 -0400
> From: "BJ Weschke" <bweschke at gmail.com>
> Subject: Re: [Asterisk-Users] Timeframe for QueueStatus values
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <79cf6330605221757g4ebfab71oaf7f27d4d47a17c2 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> On 5/22/06, mbodbg at gmx.net <mbodbg at gmx.net> wrote:
>> Hello all,
>>
>> I've a question regarding the values "completed" and "abandoned" that
>> are
>> returned by the manager command "queuestatus". What is the timeframe for
>> these values, are they counted since the last asterisk boot, or per day,
>> or
>> is the timeframe configurable?
>>
>
> Since the module was last loaded/reloaded. Therefore, there really
> isn't a "static" timeframe.
>
> --
> Bird's The Word Technologies, Inc.
> http://www.btwtech.com/
>
>
> ------------------------------
>
> Message: 15
> Date: Mon, 22 May 2006 23:23:43 -0500
> From: Greg Oliver <goliver at cistera.com>
> Subject: [Asterisk-Users] CallerID
> To: asterisk-users at lists.digium.com
> Message-ID: <1148358223.29219.2.camel at greg-laptop>
> Content-Type: text/plain
>
> I am trying to set CIDNum to nothing, but my outgoing PRI controlled by
> another PBX seems to fill in something when asterisk does not.. If I
> set a number either in the sip channel for the phone, or from
> extensions.con, it is realized.. If I try to leave them blank, or even
> Not Defined, the main number of the pri gets sent out..
>
> I am trying to debug a glitvh in or software and I need to be able to
> make a test call with unknown (blank callerid).. I can successfully set
> it to private, but that is not the same..
>
> Any ideas?
>
> TIA
>
> -Greg
>
>
>
> ------------------------------
>
> Message: 16
> Date: Mon, 22 May 2006 22:06:10 -0700
> From: Tom Lynn <tom at tomlynn.com>
> Subject: Re: [Asterisk-Users] Help Avaya 4606
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <p06572h096b3at62sd71jtla31r5fdnqri at 4ax.com>
> Content-Type: text/plain; charset=us-ascii
>
> The 4606 is a h.323 based phone. There is no SIP image to use with this
> phone.
>
> On Fri, 12 May 2006 11:11:48 -0500, you wrote:
>
>>Hello all,
>>
>>I have asterisk working well with, Sipura, but I do not manage to form
>>several phones avaya 4606, someone could have formed one avaya with
>>asterisk?
>>
>>is it possible?
>>
>>update the firmware of the phone, but I do not achieve that it registers,
>>
>>
>>I hope that someone could help me
>>
>>greetings to all
>>
>>Carlos Rojas
>
>
> ------------------------------
>
> Message: 17
> Date: Tue, 23 May 2006 15:18:07 +1000
> From: "Eric Bishop" <asterisk.eric at gmail.com>
> Subject: [Asterisk-Users] US telco lingo
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <4acda1b40605222218i4a603665tab498e1217e5e0ef at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Could someone explain to a non-US dummy the following phrases I have seen
> on
> the list.
>
> "I can provide you with tier 1 termination 6/6. I can blend or NPANXX
> breakout."
>
> "We provide US48 termination, blended rate for 1 MOU and above is .008
> with
> 6/6."
>
>
> What is 6/6?
>
> What is US48?
>
> What is blended?
>
> What is MOU?
>
> What is NPANXX breakout?
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>
> ------------------------------
>
> Message: 18
> Date: Mon, 22 May 2006 23:15:44 -0700
> From: "Kerry Garrison" <support at techdatapros.com>
> Subject: RE: [Asterisk-Users] US telco lingo
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <002a01c67e30$53862cb0$800101df at kerry01>
> Content-Type: text/plain; charset="us-ascii"
>
>
>
> Could someone explain to a non-US dummy the following phrases I have
> seen on the list.
>
> "I can provide you with tier 1 termination 6/6. I can blend or
> NPANXX breakout."
>
> "We provide US48 termination, blended rate for 1 MOU and above is
> .008 with 6/6."
>
>
> What is 6/6?
> 2/3 devil?
> Normally I would take that to be minimum 6 second billiing and
> billed in 6 seconds increments.
>
> What is US48?
> Contentinal US, lower 48, all states by Alaska and Hawaii.
>
> What is blended?
> What you do with ice, alchohol, and a mixer
>
>
>
>
>
> ------------------------------
>
> Message: 19
> Date: Tue, 23 May 2006 17:23:08 +1000
> From: Warrick Zedi <wzedi at noojeeit.com.au>
> Subject: [Asterisk-Users] Faxing - machines stop talking, line stays
> up
> To: asterisk-users at lists.digium.com
> Message-ID: <1148368988.1409.22.camel at localhost.localdomain>
> Content-Type: text/plain
>
> Hi,
>
> I haven't seen this situation described anywhere. We have an Asterisk
> server configured with 2 T4XXP cards and a TDM400P. We are using
> spandsp's txfax to send faxes on one of the E1 channels.
>
> The calls originates and the faxes start talking to each other for about
> 10 seconds. Then they just stop. The line stays up for a while after
> that until the remote end hangs up. The local line stays up for a little
> while longer and then eventually hangs up.
>
> I've fiddled with the gain settings in zapata.conf using ztmonitor as a
> guide to no avail. I've also tried busydetect=no in zapata.conf.
>
> In addition, changing the txgain appears to have no effect (going by
> ztmonitor), while rxgain does have an effect.
>
> Any suggestions welcome.
>
> Cheers,
> Warrick
>
>
>
> ------------------------------
>
> Message: 20
> Date: Tue, 23 May 2006 09:27:28 +0200
> From: "Alejandro Vargas" <alejandro.anv at gmail.com>
> Subject: Re: [Asterisk-Users] FAX and Asterisk
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <acb80a700605230027r757c5674j4c9f1326887c882 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> 2006/5/23, Rene Nelson <neririn at gmail.com>:
>>
>> I want to accept faxes via SIP/IAX2 (yes I've read the posts that it
>> isnt
>> reccomended). My PBX is 100% Virtual with the exception of one IAX
>
> I made it with iaxmodem and hylafax. It can route fax to email
> converting fax to pdf, and many things you like to do, based on the
> caller id i.e.
>
>
> --
> Alejandro Vargas
>
>
> ------------------------------
>
> Message: 21
> Date: Tue, 23 May 2006 08:35:55 +0100 (BST)
> From: John Joseph <jjk_saji at yahoo.com>
> Subject: [Asterisk-Users] TDM400P , "ztcfg ?vv error ", "Does it have
> to do with my PC hardware ?"
> To: Asterisk Users <asterisk-users at lists.digium.com>
> Message-ID: <20060523073555.97539.qmail at web34808.mail.mud.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Hi
> I have a working asterisk with TDM400P card ,
> today I was trying to install asterisk with another
> TDM400P card in another machine, I copied the
> zapata.conf and zaptel.conf of the working asterisk
> To my surprise I am getting this error
> when I run ztcfg -vv
>
> Changing signalling on channel 1 from Clear channel to
> FXS Kewlstart
> ZT_CHANCONFIG failed on channel 1: Invalid argument
> (22)
> Did you forget that FXS interfaces are configured with
> FXO signalling
> and that FXO interfaces use FXS signalling?
>
> I would like to know , whether this error is coming
> , because of my second PC hardware , I want to
> confirm really it is my PC problem , or not with my
> configuration files
> Guidance requested
> Thanks
> Joseph John
>
>
> Send instant messages to your online friends http://uk.messenger.yahoo.com
>
>
> ------------------------------
>
> Message: 22
> Date: Tue, 23 May 2006 17:40:09 +1000
> From: Warrick Zedi <wzedi at noojeeit.com.au>
> Subject: [Fwd: [Asterisk-Users] Faxing - machines stop talking, line
> stays up]
> To: asterisk-users at lists.digium.com
> Message-ID: <1148370009.1409.27.camel at localhost.localdomain>
> Content-Type: text/plain; charset="us-ascii"
>
> Sorry,
>
> I didn't mention we're using spandsp txfax (0.0.2pre25) to send the
> faxes.
> -------------- next part --------------
> An embedded message was scrubbed...
> From: Warrick Zedi <wzedi at noojeeit.com.au>
> Subject: [Asterisk-Users] Faxing - machines stop talking, line stays up
> Date: Tue, 23 May 2006 17:23:08 +1000
> Size: 4568
> Url:
> http://lists.digium.com/pipermail/asterisk-users/attachments/20060523/68c2113d/attachment-0001.eml
>
> ------------------------------
>
> Message: 23
> Date: Tue, 23 May 2006 09:58:11 +0200
> From: "Giordano Grandis" <g.grandis at invidea.it>
> Subject: [Asterisk-Users] SIP session number
> To: <asterisk-users at lists.digium.com>
> Message-ID:
> <009060C69E6D7140A8CC706BB035A9E1259DD7 at athos.tecnojest.it>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi all,
> just a question: how can i known the number of SIP session?
> In general and not for a single user.
>
> Thanks
>
> Giordano
>
> Le informazioni contenute nella presente e-mail e nei documenti
> eventualmente allegati possono essere confidenziali e sono comunque
> riservate al destinatario della stessa. La loro diffusione, distribuzione
> e/o copiatura da parte di terzi è proibita. Se avete ricevuto questa
> comunicazione per errore, Vi preghiamo di informare immediatamente il
> mittente del messaggio e di distruggere questa e-mail.
>
> This e-mail may contain confidential and/or privileged information. If you
> are not the intended recipient (or have received this e-mail in error)
> please notify the sender immediately and destroy this e-mail. Any
> unauthorised copying, disclosure or distribution of the material in this
> e-mail is strictly forbidden.
>
>
>
>
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>
> ------------------------------
>
> Message: 24
> Date: Tue, 23 May 2006 10:10:31 +0200
> From: "Asterisk" <asterisk at abraxas.si>
> Subject: [Asterisk-Users] Logger rotate & master.csv
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <7429E7E7F5F9E94D8A34C494BF5C6DACF82B at abxserver2.abraxas.si>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi guys,
>
>
>
> I have noticed that 'logger rotate' command only rotates log files in
> the /var/log/asterisk directory, but not in the subdirectories. How
> could I rotate my /var/log/asterisk/cdr-custom/Master.csv log file?
>
>
>
> Regards,
>
> Alex
>
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>
> ------------------------------
>
> Message: 25
> Date: Tue, 23 May 2006 10:11:55 +0200
> From: Kai Ober <kast.asterisk at gmx.de>
> Subject: [Asterisk-Users] Free/Open pci telco card
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> <asterisk-users at lists.digium.com>
> Message-ID: <4472C3CB.4000008 at gmx.de>
> Content-Type: text/plain; charset=ISO-8859-15; format=flowed
>
> Hi List,
>
> While I was surfing the net last week,
> I found a link for "open source" pci telco cards.
> I'm not sure if it were isdn or analog related.
>
> The layout an all the stuff was free downloadable, so that you can build
> your own cards.
>
> Does anybody have the link?
>
> Yes, I know there is google, but i searched for over an hour, but can't
> find anything.
> maybe i use the wrong search words, anny suggestions?
>
> thx
> Kay
>
>
>
> ------------------------------
>
> Message: 26
> Date: Tue, 23 May 2006 10:14:14 +0200
> From: Louis-David Mitterrand <vindex+lists-asterisk-users at apartia.org>
> Subject: [Asterisk-Users] Re: voicemail access on the Thomson ST2030 ?
> To: picciuX <matteo at picciux.it>
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <20060523081413.GA2061 at apartia.fr>
> Content-Type: text/plain; charset=us-ascii
>
> On Mon, May 22, 2006 at 12:25:34PM +0200, picciuX wrote:
>> for provisioning files to be taken, you have to change the "config_sn"
>> parameter each time you modify the file, otherwise the phone assumes
>> nothing
>> has changed.
>
> Even after a factory reset of the phone? (ie: power-cycle with
> speaker+mute buttons pressed)
>
> Thanks,
>
>
> ------------------------------
>
> Message: 27
> Date: Tue, 23 May 2006 10:21:05 +0200 (CEST)
> From: Remco Barende <asterisk at barendse.to>
> Subject: [Asterisk-Users] A call from a call file always does a
> redial?
> To: Asterisk Users List <asterisk-users at lists.digium.com>
> Message-ID: <Pine.LNX.4.62.0605231018100.5894 at raveon.vaag.nu>
> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>
>
> I have an issue with the Snom 360's (any firmware) and asterisk call
> files. When you setup a call using a call file from Asterisk and the call
> is connected, Asterisk will start to redial the call after about 5 minutes
> when the conversation is already ongoing. (Annoying and it can only be
> avoided by disabling call waiting)
>
> I tried to reproduce the problem with a GrandStream phone and a Sipura
> ATA, it doesn't occur.
>
> I guess these are 2 problems :
>
> 1) The callfile specifies that a call should not be retried, still * does
> a redial
>
> 2) I *guess* the Snom is returning a different signal than other phones
> when the call is answered up making Asterisk believe that the call
> never succeeded.
>
> I registered this as a bug in mantis previously but nobody was able to
> reproduce, I know found out that it is only happening when using a
> Snom 360 as client.
>
>
>
> ------------------------------
>
> Message: 28
> Date: Tue, 23 May 2006 10:23:15 +0200
> From: Philipp Ott <philipp.ott at avalon.at>
> Subject: Re: [Asterisk-Users] Deadlocks in 1.2.7.1
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <7CEBA694-A6A4-40DE-9A40-9DB75B1A755F at avalon.at>
> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
>
> Hello!
>
> Am 17.05.2006 um 13:05 schrieb Philipp Ott:
>
>> Hello!
>>
>> Unfortunately we are seeing lately (2-3 times during a day) that
>> asterisk seems to hang up somehow - no new calls can be made and
>> sip show peers and other commands show no obvious problem. We then
>> recompiled 1.2.7.1 with all the DEBUG_ turned on in the makefile
>> and now we see the following messages:
>>
>> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236
>> __ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc):
>> Deadlock? waited 460 sec for mutex '&iflock'?
>> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239
>> __ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor):
>> '&iflock' was locked here.
>> May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:236
>> __ast_pthread_mutex_lock: pbx.c line 2017
>> (ast_extension_state_del): Deadlock? waited 460 sec for mutex
>> '&hintlock'?
>> May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:239
>> __ast_pthread_mutex_lock: pbx.c line 1892 (ast_hint_state_changed):
>> '&hintlock' was locked here.
>> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236
>> __ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc):
>> Deadlock? waited 460 sec for mutex '&iflock'?
>> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239
>> __ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor):
>> '&iflock' was locked here.
>> May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:236
>> __ast_pthread_mutex_lock: pbx.c line 2017
>> (ast_extension_state_del): Deadlock? waited 460 sec for mutex
>> '&hintlock'?
>> May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:239
>> __ast_pthread_mutex_lock: pbx.c line 1892 (ast_hint_state_changed):
>> '&hintlock' was locked here.
>> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236
>> __ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc):
>> Deadlock? waited 460 sec for mutex '&iflock'?
>> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239
>> __ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor):
>> '&iflock' was locked here.
>> May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:236
>> __ast_pthread_mutex_lock: pbx.c line 2017
>> (ast_extension_state_del): Deadlock? waited 460 sec for mutex
>> '&hintlock'?
>> May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:239
>> __ast_pthread_mutex_lock: pbx.c line 1892 (ast_hint_state_changed):
>> '&hintlock' was locked here.
>> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236
>> __ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc):
>> Deadlock? waited 460 sec for mutex '&iflock'?
>> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239
>> __ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor):
>> '&iflock' was locked here.
>>
>> This continues until someone stops asterisks and restarts it.
>
>
> Stepping back to version 1.2.4 solves the problem of a hanging
> asterisk, however occassionally we see 5-15 seconds runs of
>
> May 23 00:28:35 DEBUG[3212] chan_sip.c: Failed to grab lock, trying
> again...
>
> messages in the log file and during this time no call processing
> happens. Then asterisk recovers from this locking state and
> continues. 1.2.7.1 hangs in there forever.
>
> Any clues as to why this happens?
>
> Regards
> Philipp Ott
>
>
>
> ------------------------------
>
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