[asterisk-users] Re: Idiot questions
On 2006-08-24 18:10:20 -0700, kritikus Araklidas [EMAIL PROTECTED] said: So: The FXO car is for the Pots lines (I.E. bellsouth line) so if you need a analog phone cennected to asterisk you need a FXS card, so if you gonna use a SIP Soft Phone (or a regular SIP Phone) you only need a network connectivity between Asterisk and SIP Phone. You don't need an FXS card. You can use ATA's and hang analog sets off of those. Or you can use phones that hook to the ethernet (SIP or IAX or?). Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Nokia E60/61/70 and SIP
On 2006-08-23 18:02:52 -0700, El Flynn [EMAIL PROTECTED] said: Hi list, Just wondering -- has anyone used the SIP phone feature on the Nokia E60/61/70 phones? We're trying to see if this would be an OK phone to get for the company, particularly since we're already running Asterisk. Not asking for a review of the phone, but rather how well the built-in SIP client works. snip I have the E60 and and although the SIP client works with asterisk, it's unreliable and not ready for prime time. There are lots of ideas for improving this, but ultimately this is yet ANOTHER VoIP product in search of a firmware update. Still, as a cell that has WIFI Voip connectivity, very promising indeed. Marty PS The phones IMAP email client can play back my asterisk voicemail messages fine too. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [Users] Mysql problem
[ Answering on both lists ] On Thu, Aug 24, 2006 at 02:52:54PM -0500, Diego Quintana Cruz wrote: Hi all, I have a problem, I can't find nowhere the asterisk-mysql package. I'm using the sarge version of yours. My repository is: deb http://rapid.sunsite.dk/rapid/ sarge main Hope you can help me find the package, I googled with no result. It got dropped of the list of packages, and hence not synced into the repository. I've just re-added it. Thanks for the catch. I've added it last night to our internal repository, a snapshot of which you can find at http://rapid.tzafrir.org.il/rapid (if you consider using that repository: it does not get extensivly tested and may break without further warning. Just as now I have uploaded a package to it and freshened it). We'll refresh the main repository on Sunday after some testings. (There are some other pending fixes) -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Idiot questions
Hello, If you can buy a TDM400, good! Support Digium Cheers, Madhawa Nilesh Londhe wrote: I would suggest buying a very low price FXO to begin with which would probably be x100p PCI card at ebay for about $10 +shipping. On 8/24/06, *Adam Collard* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You will need a TDM400 with an FXO module for each line you want. A TDM400 supports up to four lines or analog stations. For two lines, you should get a TDM04B. -Original message- From: joea, j4computers [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Date: Thu, 24 Aug 2006 14:58:21 -0700 To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Idiot questions As a complete newcomer to Asterisk, Digium and PBX, I have several questions. But I'll start with this. To setup a simple system with only a couple of POTS lines, I gather I will need a TDM400 board with FXO and/or FXS modules. So, a TDM400 card will support up to two analog (POTS) lines? joea ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Adam Collard President Digital Telecom of Michigan, Inc. [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (517) 233-1072 Direct Office (800) 420-3803 x4101 Office (517) 766-5902 Fax This email may be confidential. Any distribution, use or copying of this email or the information it contains by other than an intended recipient is unauthorized. If you received this email in error, please advise me (by return email or otherwise) immediately. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IP phone with 2 ethernet jacks
Hi, Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wanted Sipura but they don't have such product. Thanks, Mindaugas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] quadBRI beronet card: how to specify which ISDN channel to use to make calls
Hi, I have a quadBRI beronet ISDN card. Is there anybody who knows how to choose the channel to make calls? I tried with Dial(mISDN/1-1/) to choose channel 1 of port 1 but without success. TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemailmain
On Thu, Aug 24, 2006 at 04:08:01PM -0400, existx wrote:
Howdy,
I have a Debian box using Debian's Asterisk package.
Just to be clear about the version: I assume that the version is:
http://packages.debian.org/stable/comm/asterisk
(1:1.0.7.dfsg.1-2sarge3 or 1:1.0.7.dfsg.1-2)
If you don't lack disk space on that system, than install the package
asterisk-doc . It will install a huge pile of unnecessary API docs. But
also /usr/share/doc/asterisk-doc/examples with the sample configs.
People can leave
voicemail for the extensions that are setup in the configuration, and
asterisk e-mail's the user a .wav file (voicemail.conf). This works
perfect.
However, I want to have VoicemailMain sit on an extension so people
can call in, change their greeting, listen too voicemail, etc.
extensions.conf:
exten = 2999,1,Answer
exten = 2999,2,Wait,2
exten = 2999,3,Voicemailmain()
My understand is, that this should allow any user to call up. Enter in
their mailbox number (currently the same as their extension) and
password. However, I cannot dial this extension after reloading
asterisk.
This is normally an issue with detecting the DTMFs in the call. What
phones are the users using? How are they connected to Asterisk?
If those are SIP phones, then both sterisk and the phones need to agree
on the DTMF encoding method. See the dtmfmode option in sip.conf.
(Note that 1.0 does not have dtmfmode=auto)
Also: VoicemailMain can take a argument for a username. Usually the
caller's caller ID will also match its mailbox number (at least for
internal calls). In such a case you can use the following hack:
exten = _299[89],1,Answer
exten = _299[89],2,Wait,2 ; try waiting just 1?
exten = _2998,3,Voicemailmain(s${CALLERIDNUM})
exten = _2999,3,Voicemailmain()
(Note that this is asterisk 1.0 syntax. In Asterisk 1.2 use
Voicemailmain(${CALLERID(num)},s)
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406 jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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RE: [asterisk-users] Re: column width in CLI
I think he actually needs show channels verbose *CLI help show channels Usage: show channels [concise|verbose] Lists currently defined channels and some information about them. If 'concise' is specified, the format is abridged and in a more easily machine parsable format. If 'verbose' is specified, the output includes more and longer fields. Cheers SKM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Sent: Wednesday, August 23, 2006 6:59 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: column width in CLI Try show channels concise -- -- Steven http://www.glimasoutheast.org Shaun Hofer [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, Can the column width for commands run in the Asterisk CLI be increased? Currently when I run 'show channels' I can't see the whole channels id/name as its to long for the columns width and is cut off. I need to grab a list of active channels, which is currently not do able. Thanks Shaun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Adding/Removing Prefixes
I now need to remove the 9 but then prefix another number onto
the phone number before dialing now but am unsure how to do
this is the dialplan.
Simple...for instance, if you wish to prefix 123 before the number just do:
Dial(SIP/123${EXTEN}
Would someone be able to point me in the right direction or
provide an example diaplan that does this?
Many Thanks in Advance
SP
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RE: [asterisk-users] How to set externip in sip.conf automatically?
I believe you want to use ${ENV(variable)}.. From asterisk's CLI:
*CLIshow function ENV
-= Info about function 'ENV' =-
[Syntax]
ENV(envname)
[Synopsis]
Gets or sets the environment variable specified
Note that ENV is a function...you need to encase the argument inside
parentheses
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Larry Alkoff
Sent: Wednesday, August 23, 2006 9:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to set externip in sip.conf
automatically?
As stated in the original post, when I entter the IP with an
editor directly into sip.conf calls work just fine but I am
looking for a way to have that done _automatically_.
The Asterisk - Future of Telephony book says it is possible
for Asterisk to access a Linux environment variable containing
the IP information in the form of ${ENV{variable}}.
It doesn't seem to work. I am asking how to make it work.
Larry
Watkins, Bradley wrote:
If you already have the IP in a file, why don't you set it up so the
file itself says: externip=xx.xx.xx.xx and then do a #include in
sip.conf for the /etc/myip file? I believe you'll have to do a sip
reload either way (which can obviously be part of your cron job) if
you're not already, but that should do what you're looking to do.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Larry
Alkoff
Sent: Tuesday, August 22, 2006 9:34 PM
To: Asterisk-users; Austin-asterisk-users
Subject: [asterisk-users] How to set externip in sip.conf
automatically?
I need to give Asterisk access to my external IP address
to prevent
the NAT problem where caller cannot hear the callee's voice.
According to Asterisk - The Future of Telephony page 92 Environment
Variables:
Environment variables are a way of accessing Unix environment
variables from within Asterisk. They are referenced in the form of
${ENV{var}}
where var is the Unix environment variable you wish to reference.
My external IP is placed each night in a file call /etc/myip and
placed in the $MYIP variable by /etc/bashrc when an shell is loaded.
So I have /etc/myip refreshed each night in a cron job and when a
shell is opened /etc/bashrc does:
export MYIP=`cat /etc/myip`
To access the variable in sip.conf I have tried:
externip=${ENV(EXTERNIP)}
and
${ENV($EXTERNIP)}
but neither seems to work.
Is this the correct syntax? Did I misinterpret the book?
I say neither seems to work because When I hard code
externip=69.91.84.176
there are no NAT problems but when I try to access the $MYIP
variable
either of the ways above NAT prevents me hearing the callee's voice.
I have tried but not found a way to directly access the contents of
MYIP to the console using the CLI. Is there a way to see or
set _any_
Linux enviromnent variable using the CLI? More generally, how do I
access the Linux shell from the CLI?
The problem with simply using
externip=69.91.94.176
is that number is subject to change and I don't know an easy way to
automatically write the value into sip.conf programatically.
I could have just said how do I do this but wanted to show
that I've
done my homework.
Thanks for any help.
Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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RE: [asterisk-users] IP phone with 2 ethernet jacks
We like the SNOM 360 Phones. They have really good features. Guido -Ursprüngliche Nachricht- Von: Mindaugas Kuprys [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 25. August 2006 09:40 An: asterisk-users Betreff: [asterisk-users] IP phone with 2 ethernet jacks Hi, Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wanted Sipura but they don't have such product. Thanks, Mindaugas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] MySQL CDR
Download the asterisk-addons package. It contains several addons, including all the mysql additions. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Quintana Cruz Sent: Thursday, August 24, 2006 4:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] MySQL CDR Hi everyone, I finished installing the Xorcom Rapid's Asterisk Packages with amportal (1.10.10), but i wasn't able to find the asterisk-mysql package. Any idea what happened there?, Is there another reposiitory for that package for asterisk 1.0.11. Or could somebody send me the cdr_addon_mysql.so file? Thanks for your responses, -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk t38passthrough
I have enabled in ATA's GUI the T.38 codec as the preferred codec, but of course if it detects that the other side doesn't work with T.38, it tries with the following codec preferences like G.711. On the other side there is PSTN, as I deliver my traffic in IP to a Telco that uses also T.38. The fact is that I think Asterisk-t38 branch installation isn't doing T.38 bypass... If it were, G.711 wouldn't be used... I guess... Any help? Ricardo. Edgar Barbosa wrote: Also, make sure you have a T.38 enabled device at the other end… Edgar *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *William Piper *Sent:* quinta-feira, 24 de Agosto de 2006 21:09 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk t38passthrough Perhaps a stupid suggestion... but did you make sure that the ATA had the T38 selected in the GUI? bp On 8/24/06, *Ricardo Carvalho* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I've installed Asterisk t38passthrough branch and I'm using one Grandstream ATA to connect Asterisk to a Fax machine. Every time I send a fax, it gets sent using codec G711, and never T.38. I added the following parameters in the [general] section as well as in device configurations: t38pt_udptl = yes t38pt_rtp = yes t38pt_tcp = yes I think that's the only thing that is needed to do to enable T.38 pass through... Why does Asterisk keeps sending in G711? Any help? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Annoying Bristuff
Its working for me with no errors. * 1.2.10 bristuff 0.3.0-pre1s with kernel 2.6.15.4. My setup is kind of "special" as its build with Openembedded and runs from a CF on a [EMAIL PROTECTED] Recently i was able to port *+bristuff + zaptel to an embedded powerpc platform and works there also without any major issues. Why don't you trya 2.6 kernel maybe the problem is there (unlikely though) Stelios HiCan anyone confirm a working asterisk 1.2 from bristuff with 1 port PCI, hfc-s based ISDN card (zaphfc driver). If so, could you send your configuration. I mean OS (linux distribution) type, kernel version.Thanks in advanceCheersAndrew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk t38passthrough
Ricardo Carvalho schrieb: ... tries with the following codec preferences like G.711. On the other side there is PSTN, as I deliver my traffic in IP to a Telco that uses also Hi, that is not passthrough! You will need something to translate T.38 to one of the ordinary fax/modem-modulations, when switching to PSTN. Imho, this is not and will never be handled by asterisk's T.38 passthrough support. Anyway, Steve Underwood started to implement some T.38 support for his packages (spandsp/rxfax/txfax). Imho, this is, what you'll need. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Trunk with multiple IPs?
I wish I could offer some direct help on whether or not your method with a comma separated list would work, but I can't. However, you could always create a few entries using different formats and then run some tests against them -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Lawetz Sent: Wednesday, August 23, 2006 9:42 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Trunk with multiple IPs? Still no answers huh? I've asked a couple of time how to do this, and by the lack of answers, I'm guessing there is no way. The workaround unfortunately is to create an entry for each IP address in the range (I hope you don't have to open up a whole C class) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Warren (mailing lists) Sent: August 22, 2006 3:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Trunk with multiple IPs? How do I enter a trunk with multiple IPs. xyz voip provider has 4 IPs and I want to allow incoming from any of them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4 Do I put 4 separate host= lines, do I put a single host=line that is comma separated or do I have to set up 4 separate incoming trunks? TIA, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does anyone use T.38?
Does anyone use T.38 for fax? If you use it, what hardware / software do you use? Thanks, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Annoying Bristuff
sorry for the html post :( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] quintum Calling Card
Hello Jonathan,I tried in quintum to route my server with any dialed number. but i am not agble to get in quintum FXO line configuration, so i can route the call to my asterisk.do u have any about quintum how i can route calls to server once FXO line will be called?Abdul Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] quintum Calling Card
Ive only used a Quintum a few times,sorry. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Abdul Sent: Friday, August 25, 2006 6:49 AM To: Asterisk-Users@lists.digium.com Subject: RE: [asterisk-users] quintum Calling Card Hello Jonathan, I tried in quintum to route my server with any dialed number. but i am not agble to get in quintum FXO line configuration, so i can route the call to my asterisk. do u have any about quintum how i can route calls to server once FXO line will be called? Abdul Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple Vulnerabilities in Asterisk 1.2.10 (Fixed in 1.2.11)
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
From: http://www.sineapps.com/news.php?rssid=1448
MuLabs has posted details of multiple vulnerabilities in Asterisk 1.2.10.
Excerpt:
Vulnerability Details:
A remote stack buffer overflow condition in Asterisk's MGCP
implementation could allow for arbitrary code execution. The vulnerable
code is triggered with the use of a malformed AUEP (audit endpoint)
response message.
A second issue exists in the handling of file names sent to the
Record()application which could lead to arbitrary code execution via a
format string attack or arbitrary file-overwrite via directory traversal
techniques. The impact of this vulnerability is minimal, however, as it
requires an administrator to use a client-controlled variable as part of
the filename.
Solution:
Mu Security would like to thank the Asterisk security team for their
timely response to these issues.
A patch for the buffer overflow is available from the following link:
http://ftp.digium.com/pub/asterisk/asterisk-1.2.11-patch.gz
To protect against the Record() vulnerability, do not use
user-controlled variables ( eg, ${CALLERIDNAME} ) as part of the the
filename argument.
- --
Cheers,
Matt Riddell
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
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[asterisk-users] How do you simultaniously dial multiple MSNs on one ISDN BRI b-channel?
Hello, I'm fairly new to Asterisk. Installation went fine, and things seem to work, but I have 1 problem. Hardware: 2 HFC ISDN cards (1 in TE mode and 1 in NT mode) 1 SIP On the inside (NT mode card) I have 3 ISDN phones. Everything is connected with all cables and extra resistors, and all 3 phones can dial and be dialled. When I try to dial all 3 phones simultaniously, with Dial(Zap/g2/10Zap/g2/11Zap/g2/12,60,m(default)t) then 2 phones ring and the last one is busy/congestion. I assume its cause I only have 2 b-channels. How do I make all 3 phones ring using only 1 channel? It can be done. I also have a hardware PBX (Elmeg C46) which does that now. Can anyone help me how to do it in Asterisk? -- Best regards, Henrik Woffinden ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Trunk with multiple IPs?
Hi, I've only just now seen this post. This is how we have setup. In sip.conf [xxx.xxx.xx1] host = xxx.xxx.xx1 type = friend insecure = very context = your-context canreinvite=no [xxx.xxx.xx2] host = xxx.xxx.xx2 type = friend insecure = very context = your-context canreinvite=no [xxx.xxx.xx3] host = xxx.xxx.xx3 type = friend insecure = very context = your-context canreinvite=no [xxx.xxx.xx4] host = xxx.xxx.xx4 type = friend insecure = very context = your-context canreinvite=no Hope this helps. Paulo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rushowr Sent: Friday, August 25, 2006 4:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Trunk with multiple IPs? I wish I could offer some direct help on whether or not your method with a comma separated list would work, but I can't. However, you could always create a few entries using different formats and then run some tests against them -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Lawetz Sent: Wednesday, August 23, 2006 9:42 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Trunk with multiple IPs? Still no answers huh? I've asked a couple of time how to do this, and by the lack of answers, I'm guessing there is no way. The workaround unfortunately is to create an entry for each IP address in the range (I hope you don't have to open up a whole C class) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Warren (mailing lists) Sent: August 22, 2006 3:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Trunk with multiple IPs? How do I enter a trunk with multiple IPs. xyz voip provider has 4 IPs and I want to allow incoming from any of them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4 Do I put 4 separate host= lines, do I put a single host=line that is comma separated or do I have to set up 4 separate incoming trunks? TIA, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trunk with multiple IPs?
I don't believe that addresses the OP's original post since he was talking about limiting incoming calls from specific IP addresses. You might want to validate how secure your definitions are considering the type=friend approach. Lists @ EMS wrote: Hi, I've only just now seen this post. This is how we have setup. In sip.conf [xxx.xxx.xx1] host = xxx.xxx.xx1 type = friend insecure = very context = your-context canreinvite=no [xxx.xxx.xx2] host = xxx.xxx.xx2 type = friend insecure = very context = your-context canreinvite=no [xxx.xxx.xx3] host = xxx.xxx.xx3 type = friend insecure = very context = your-context canreinvite=no [xxx.xxx.xx4] host = xxx.xxx.xx4 type = friend insecure = very context = your-context canreinvite=no Hope this helps. Paulo I wish I could offer some direct help on whether or not your method with a comma separated list would work, but I can't. However, you could always create a few entries using different formats and then run some tests against them Still no answers huh? I've asked a couple of time how to do this, and by the lack of answers, I'm guessing there is no way. The workaround unfortunately is to create an entry for each IP address in the range (I hope you don't have to open up a whole C class) -Original Message- How do I enter a trunk with multiple IPs. xyz voip provider has 4 IPs and I want to allow incoming from any of them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4 Do I put 4 separate host= lines, do I put a single host=line that is comma separated or do I have to set up 4 separate incoming trunks? TIA, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Annoying Bristuff
On 8/25/06, Stelios Koroneos [EMAIL PROTECTED] wrote: Its working for me with no errors. * 1.2.10 bristuff 0.3.0-pre1s with kernel 2.6.15.4. My setup is kind of special as its build with Openembedded and runs from a CF on a [EMAIL PROTECTED] Recently i was able to port *+bristuff + zaptel to an embedded powerpc platform and works there also without any major issues. Why don't you trya 2.6 kernel maybe the problem is there (unlikely though) SteliosHi Thanks for your reply I will switch to 2.6 kernel asap and of course send the results to the list. Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Parking Ring Back (Snoms)
You can have it come back on another line appearance that is set with different ringtone. On 8/24/06, J. Oquendo [EMAIL PROTECTED] wrote: Quick question maybe someone can point me in the right direction... Caller -- Receptionist -- ParksCall Receptionist makes announcement for individual to pick up parked call. No one picks up so it rings back to receptionist within a minute and a half. Is there any way to change the ringer for a parked call coming back since their call wasn't answered? -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom phones locking up
For what its worth, even a Cisco Switch can go bad or be setup wrong. I would also disable the network sensing on the phones, can help. On 8/24/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi All I have had a problem with a few Snom 320's on several sites locking up after a few days. I am running application ver 6.2.2 with the latest jffs2 ver and tried the latest 5.x ver with similar results. Is this also experienced with other Snom users? I know some posts say it could be the network switches etc, but Cisco? I fail to see how a switch could bring down a device. Kind Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom microbrowser issue Error HTTP 406 withIIS
Title: Message Thanks, but we have reasons to want to make it work with IIS. Anyone have a hint of what is the issue? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas GarstangSent: Thursday, August 24, 2006 6:46 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Polycom microbrowser issue Error HTTP 406 withIIS We had a similar problem. Eventuallywe gave up and just used apache. We found that _exactly_ the same content would not work with IIS, but WOULD work with Apache. -Original Message-From: Phil Menico [mailto:[EMAIL PROTECTED]Sent: Thursday, August 24, 2006 3:06 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Polycom microbrowser issue Error HTTP 406 with IIS I have no where else to turn to so if anyone has an answer please send it. I am running sip version 1.6.on a Polycom 601on Asterisk and am unable to get the microbroser to work. The phone returns a 406 error for both idle and services. I can see the file being requested and the subsequent 406 error in the IIS log files. Any ideas on what permissions are needed in IIS or how to format the webpage file? I tried both these 2 files with no luck XHTML file 1: html head /head body Hello phil post /body/html XHTML file 2: ?xml version="1.0" encoding="UTF-8"?html xmlns="http://www.w3.org/1999/xhtml" xml:lang="en" lang="en" head titleVirtual Library/title /head body PHello phil/P /body/html Log info from IIS: 2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET /Polycom/ - 302 0 295 202 0 HTTP/1.1 10.0.1.210:81 Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 - -2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET /Polycom/post.htm - 406 0 4085 242 10 HTTP/1.1 10.0.1.210:81 Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 - http://10.0.1.210:81/Polycom Thank you. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime and hints
Ok. So what is the problem ? - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 24, 2006 12:17 PM Subject: RE: [asterisk-users] Realtime and hints But... you need _both_ in your dialplan. My extensions.conf has: exten = 2944054,hint, SIP/2944054 exten = 2944054,1, Dial(SIP/2944054) ie two lines for the hint. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints Doug I have Exten = 10,hint,SIP/11010 and in mysql I have exten = 10,1,Dial(SIP/11010) - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 21, 2006 3:37 PM Subject: [asterisk-users] Realtime and hints Can realtime be used with hints? How would you get the following into the database given that the priority column is numeric, and that there is no application for the first entry? exten = 2944006,hint,SIP/2944006 exten = 2944006,1,Dial(SIP/2944006) Every time I touch realtime I hit obstacles. How are others getting around this limitation? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime and hints
I dont know why it is working but it is. My first line I have in extensions.conf and the second I have in MySql. - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 24, 2006 1:00 PM Subject: RE: [asterisk-users] Realtime and hints I don't see how that helps. If you have a portion of the hint still in extensions.conf, then what use is the database? -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Realtime and hints That's what he was gettin at. Take the second line out, and put the first priority in the database. On Thu, 2006-08-24 at 10:17 -0600, Douglas Garstang wrote: But... you need _both_ in your dialplan. My extensions.conf has: exten = 2944054,hint, SIP/2944054 exten = 2944054,1, Dial(SIP/2944054) ie two lines for the hint. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints Doug I have Exten = 10,hint,SIP/11010 and in mysql I have exten = 10,1,Dial(SIP/11010) - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 21, 2006 3:37 PM Subject: [asterisk-users] Realtime and hints Can realtime be used with hints? How would you get the following into the database given that the priority column is numeric, and that there is no application for the first entry? exten = 2944006,hint,SIP/2944006 exten = 2944006,1,Dial(SIP/2944006) Every time I touch realtime I hit obstacles. How are others getting around this limitation? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [RESOLVED] One way audion on Sangoma
I wasn't carefull enough and in my configs I had echo cancelation enabled. For some reason it worked no problem when I was using Zaptel 1.2.7 . It only acted up as soon as I installed 1.2.8. As soon as disabled echo cancelation it started working like a charm. Dovid - Original Message - From: Dovid Bender To: asterisk-users@lists.digium.com Sent: Wednesday, August 23, 2006 7:06 PM Subject: [asterisk-users] One way audion on Sangoma Hi List, I have an A200 with echo can. 2-FXO and 2 FXS. Today I went and upgraded asterisk, zaptel and libpri. I ran the sangoma util to patch asterisk. When I started up asterisk ZAP1 worked like a charm. However ZAP2 has been acting up. I only get one way audio on it. The person that I call can hear me however I can not hear them at all. I tired switching around the lines but to no avail. It seems that only zap2 is giving the problems. Anyone have any suggestions ? Can it be that ZAP2 just crapped out today or does it have to do with the upgrade. I also want to mention that I didnt use the system all day so I dont know if it was working earlier (before I upgraded asterisk) or not. Thanks. Dovid ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can i use the FXO of a addpack in Asterisk
Is it possible to use a fxo channel of a addpack voip route as incomming channel voor asterisk? Or are there other external fxo channels that you can use for asterisk. Thanks Han ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [RESOLVED] One way audion on Sangoma
On 09:19, Fri 25 Aug 06, Dovid Bender wrote: I wasn't carefull enough and in my configs I had echo cancelation enabled. For some reason it worked no problem when I was using Zaptel 1.2.7 . It only acted up as soon as I installed 1.2.8. As soon as disabled echo cancelation it started working like a charm. The sangoma has hardware echo cancel ? If so it makes sence, because the settings in zapata.conf are for the software echo cancel, and that should be disabled for all interfaces that have hardware echo can. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Singapore
Hi is there anyone on the list who is installing Asterisk in Singapore (or has installed servers in a hosted facility in Singapore?) As an alternative Id also like to talk to anyone doing similar in Malaysia (though this is a backup). If so can you please email me with your contact details. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [RESOLVED] One way audion on Sangoma
The sangoma has hardware echo cancel ? If so it makes sence, because the settings in zapata.conf are for the software echo cancel, and that should be disabled for all interfaces that have hardware echo can. No, that is incorrect. From http://wiki.sangoma.com/wanpipe-asterisk-configure: The Wanpipe TDM driver enables HW Echo Cancellation only on channels that have active calls: It waits for zaptel to enable echo cancellation after the call has been established. Therefore, Echo Cancellation option MUST be enabled in /etc/asterisk/zapata.conf. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Idiot questions
Nilesh Londhe wrote: I would suggest buying a very low price FXO to begin with which would probably be x100p PCI card at ebay for about $10 +shipping. On 8/24/06, *Adam Collard* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You will need a TDM400 with an FXO module for each line you want. A TDM400 supports up to four lines or analog stations. For two lines, you should get a TDM04B. -Original message- From: joea, j4computers [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Date: Thu, 24 Aug 2006 14:58:21 -0700 To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Idiot questions As a complete newcomer to Asterisk, Digium and PBX, I have several questions. But I'll start with this. To setup a simple system with only a couple of POTS lines, I gather I will need a TDM400 board with FXO and/or FXS modules. So, a TDM400 card will support up to two analog (POTS) lines? joea ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Adam Collard President Digital Telecom of Michigan, Inc. [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (517) 233-1072 Direct Office (800) 420-3803 x4101 Office (517) 766-5902 Fax This email may be confidential. Any distribution, use or copying of this email or the information it contains by other than an intended recipient is unauthorized. If you received this email in error, please advise me (by return email or otherwise) immediately. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users some of those x100p cards have issues with echo.. definitely get the tdm card if you can signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Singapore
Hello, This is not a correct list. try biz list. Cheers, Madhawa Dean Collins wrote: Hi is there anyone on the list who is installing Asterisk in Singapore (or has installed servers in a hosted facility in Singapore?) As an alternative I’d also like to talk to anyone doing similar in Malaysia (though this is a backup). If so can you please email me with your contact details. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]+1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP phone with 2 ethernet jacks
On Friday 25 August 2006 03:39, Mindaugas Kuprys wrote: Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wanted Sipura but they don't have such product. Polycom IP501. Failing that, IP430. If you want to go for the gold, go IP601. I heart polycom. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Idiot questions
I actually had a look at one on ebay. What concerned me was the fact that the seller had set it on a carpet for the pictures. I was concerned about static damage. Too late now, tho. I'm still conflicted about it. Sigh. joea Dualcall.com[EMAIL PROTECTED] wrote on 8/25/2006 2:43 AM: Hello, If you can buy a TDM400, good! Support Digium Cheers, Madhawa Nilesh Londhe wrote: I would suggest buying a very low price FXO to begin with which would probably be x100p PCI card at ebay for about $10 +shipping. On 8/24/06, *Adam Collard* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You will need a TDM400 with an FXO module for each line you want. A TDM400 supports up to four lines or analog stations. For two lines, you should get a TDM04B. -Original message- From: joea, j4computers [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Date: Thu, 24 Aug 2006 14:58:21 -0700 To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Idiot questions As a complete newcomer to Asterisk, Digium and PBX, I have several questions. But I'll start with this. To setup a simple system with only a couple of POTS lines, I gather I will need a TDM400 board with FXO and/or FXS modules. So, a TDM400 card will support up to two analog (POTS) lines? joea ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Adam Collard President Digital Telecom of Michigan, Inc. [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (517) 233-1072 Direct Office (800) 420-3803 x4101 Office (517) 766-5902 Fax This email may be confidential. Any distribution, use or copying of this email or the information it contains by other than an intended recipient is unauthorized. If you received this email in error, please advise me (by return email or otherwise) immediately. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anybody using Eicon SoftIP with Asterisk
Hi I too try run Diva Soft IP and Asterisk, but I have problem. All time diva is busy. In Diva monitor I see incoming call, but error occur. Have you or somebody any idea? -- 15:52:39.062 I 5 LC-VerifySoftipLicence 15:52:39.062 I 5 LC-chdir to C:\Program Files\Diva Server\Licenses 15:52:39.062 I 5 LC-Current AKI File: License000.lic 15:52:39.062 I 5 LC-CertFile: C:\Program Files\Diva Server\Licenses/akirootcert.pem 15:52:39.187 I 5 LC-Check DUID: SAAL66AOMJEYAAQ 15:52:39.187 I 5 LC-Check_AKID_PPCID Passed 15:52:42.265 R 5 I-diva_hssua_listenToNetwork: 1 15:52:42.265 V 5 V-ENTER initEvent 15:52:42.265 V 5 V-initEvent 1 15:52:42.265 V 5 V-initEvent 3 15:52:42.265 V 5 V-initEvent 4 15:52:42.265 V 5 V-return initEvent 15:52:42.265 n 5 SN-RX UDP frame 15:52:42.265 c 5 SM- 15:52:42.265 c 5 SM-SIPR begin from IP:192.168.0.150 PORT:5060 15:52:42.265 c 5 SM- INVITE sip:192.168.0.57 SIP/2.0 15:52:42.265 c 5 SM- Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK3a544ef1;rport 15:52:42.265 c 5 SM- From: domin sip:[EMAIL PROTECTED];tag=as2b16c84d 15:52:42.265 c 5 SM- To: sip:192.168.0.57 15:52:42.265 c 5 SM- Contact: sip:[EMAIL PROTECTED] 15:52:42.265 c 5 SM- Call-ID: [EMAIL PROTECTED] 15:52:42.265 c 5 SM- CSeq: 102 INVITE 15:52:42.265 c 5 SM- User-Agent: Asterisk PBX 15:52:42.265 c 5 SM- Max-Forwards: 70 15:52:42.265 c 5 SM- Date: Fri, 25 Aug 2006 13:36:16 GMT 15:52:42.265 c 5 SM- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY 15:52:42.265 c 5 SM- Content-Type: application/sdp 15:52:42.265 c 5 SM- Content-Length: 208 15:52:42.265 c 5 SM- 15:52:42.265 c 5 SM- v=0 15:52:42.265 c 5 SM- o=root 3251 3251 IN IP4 192.168.0.150 15:52:42.265 c 5 SM- s=session 15:52:42.265 c 5 SM- c=IN IP4 192.168.0.150 15:52:42.265 c 5 SM- t=0 0 15:52:42.265 c 5 SM- m=audio 10338 RTP/AVP 0 8 4 15:52:42.265 c 5 SM- a=rtpmap:0 PCMU/8000 15:52:42.265 c 5 SM- a=rtpmap:8 PCMA/8000 15:52:42.265 c 5 SM- a=rtpmap:4 G723/8000 15:52:42.265 c 5 SM- a=silenceSupp:off - - - - 15:52:42.265 B 5 253 bytes 0x 49 4E 56 49 54 45 20 73 69 70 3A 31 39 32 2E 31 INVITE sip:192.1 0x0010 36 38 2E 30 2E 35 37 20 53 49 50 2F 32 2E 30 0D 68.0.57 SIP/2.0. 0x0020 0A 56 69 61 3A 20 53 49 50 2F 32 2E 30 2F 55 44 .Via: SIP/2.0/UD 0x0030 50 20 31 39 32 2E 31 36 38 2E 30 2E 31 35 30 3A P 192.168.0.150: 0x0040 35 30 36 30 3B 62 72 61 6E 63 68 3D 7A 39 68 47 5060;branch=z9hG 0x0050 34 62 4B 33 61 35 34 34 65 66 31 3B 72 70 6F 72 4bK3a544ef1;rpor 0x0060 74 0D 0A 46 72 6F 6D 3A 20 22 64 6F 6D 69 6E 22 t..From: domin 0x0070 20 3C 73 69 70 3A 64 6F 6D 69 6E 40 31 39 32 2E sip:[EMAIL PROTECTED] 0x0080 31 36 38 2E 30 2E 31 35 30 3E 3B 74 61 67 3D 61 168.0.150;tag=a 0x0090 73 32 62 31 36 63 38 34 64 0D 0A 54 6F 3A 20 3C s2b16c84d..To: 0x00a0 73 69 70 3A 31 39 32 2E 31 36 38 2E 30 2E 35 37 sip:192.168.0.57 0x00b0 3E 0D 0A 43 6F 6E 74 61 63 74 3A 20 3C 73 69 70 ..Contact: sip 0x00c0 3A 64 6F 6D 69 6E 40 31 39 32 2E 31 36 38 2E 30 :[EMAIL PROTECTED] 0x00d0 2E 31 35 30 3E 0D 0A 43 61 6C 6C 2D 49 44 3A 20 .150..Call-ID: 0x00e0 32 61 35 63 64 37 34 65 31 38 38 35 30 37 33 33 2a5cd74e18850733 0x00f0 34 30 37 62 38 39 31 66 33 63 65 33 65 407b891f3ce3e 15:52:42.265 B 5 253 bytes 0x 39 64 66 40 31 39 32 2E 31 36 38 2E 30 2E 31 35 [EMAIL PROTECTED] 0x0010 30 0D 0A 43 53 65 71 3A 20 31 30 32 20 49 4E 56 0..CSeq: 102 INV 0x0020 49 54 45 0D 0A 55 73 65 72 2D 41 67 65 6E 74 3A ITE..User-Agent: 0x0030 20 41 73 74 65 72 69 73 6B 20 50 42 58 0D 0A 4D Asterisk PBX..M 0x0040 61 78 2D 46 6F 72 77 61 72 64 73 3A 20 37 30 0D ax-Forwards: 70. 0x0050 0A 44 61 74 65 3A 20 46 72 69 2C 20 32 35 20 41 .Date: Fri, 25 A 0x0060 75 67 20 32 30 30 36 20 31 33 3A 33 36 3A 31 36 ug 2006 13:36:16 0x0070 20 47 4D 54 0D 0A 41 6C 6C 6F 77 3A 20 49 4E 56 GMT..Allow: INV 0x0080 49 54 45 2C 20 41 43 4B 2C 20 43 41 4E 43 45 4C ITE, ACK, CANCEL 0x0090 2C 20 4F 50 54 49 4F 4E 53 2C 20 42 59 45 2C 20 , OPTIONS, BYE, 0x00a0 52 45 46 45 52 2C 20 53 55 42 53 43 52 49 42 45 REFER, SUBSCRIBE 0x00b0 2C 20 4E 4F 54 49 46 59 0D 0A 43 6F 6E 74 65 6E , NOTIFY..Conten 0x00c0 74 2D 54 79 70 65 3A 20 61 70 70 6C 69 63 61 74 t-Type: applicat 0x00d0 69 6F 6E 2F 73 64 70 0D 0A 43 6F 6E 74 65 6E 74 ion/sdp..Content 0x00e0 2D 4C 65 6E 67 74 68 3A 20 32 30 38 0D 0A 0D 0A -Length: 208 0x00f0 76 3D 30 0D 0A 6F 3D 72 6F 6F 74 20 33 v=0..o=root 3 15:52:42.265 B 5 195 bytes 0x 32 35 31 20 33 32 35 31 20 49 4E 20 49 50 34 20 251 3251 IN IP4 0x0010 31 39 32 2E 31 36 38 2E 30 2E 31 35 30 0D 0A 73 192.168.0.150..s 0x0020 3D 73 65 73 73 69 6F 6E 0D 0A 63 3D 49 4E 20 49 =session..c=IN I 0x0030 50 34 20 31 39 32 2E 31 36 38 2E 30 2E 31 35 30 P4 192.168.0.150 0x0040 0D 0A 74 3D 30 20 30 0D 0A 6D 3D 61
RE: [asterisk-users] Realtime and hints
I seem to remember someone posting somewhere (was it the list or some site I was browsing...) where someone had created a hint lookup table and just put a db lookup in the dialplan for the hint priority. Then you just need one line to cover all your hints and the db will handle everything else. You just have to figure out the best realtime - static combination that works for you, just like ALL of us have had to. Trial and error does wonders. Aaron On Thu, 2006-08-24 at 11:00 -0600, Douglas Garstang wrote: I don't see how that helps. If you have a portion of the hint still in extensions.conf, then what use is the database? -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Realtime and hints That's what he was gettin at. Take the second line out, and put the first priority in the database. On Thu, 2006-08-24 at 10:17 -0600, Douglas Garstang wrote: But... you need _both_ in your dialplan. My extensions.conf has: exten = 2944054,hint, SIP/2944054 exten = 2944054,1, Dial(SIP/2944054) ie two lines for the hint. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints Doug I have Exten = 10,hint,SIP/11010 and in mysql I have exten = 10,1,Dial(SIP/11010) - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 21, 2006 3:37 PM Subject: [asterisk-users] Realtime and hints Can realtime be used with hints? How would you get the following into the database given that the priority column is numeric, and that there is no application for the first entry? exten = 2944006,hint,SIP/2944006 exten = 2944006,1,Dial(SIP/2944006) Every time I touch realtime I hit obstacles. How are others getting around this limitation? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Parking Ring Back (Snoms)
Assume 4XX extensions and the SNOMs have a page extension that auto
answers in the 5XX range to match.
; Parking
exten = 6,1,NoOp()
exten =
6,n,ParkAndAnnounce(call:ha/on:PARKED|105|SIP/5${BLINDTRANSFER:5:2}|default,74${BLINDTRANSFER:5:2},1)
exten = 6,hint,Local/6
; Parking Ring back
exten = _74XX,1,Set(CALLERID(name)=Parked Call)
exten = _74XX,n,ChanIsAvail(SIP/${EXTEN:1}|sj)
exten = _74XX,n,Dial(SIP/${EXTEN:1}|30)
exten = _74XX,n,Goto(default,${EXTEN},102)
exten = _74XX,102,Goto(operator,s,1)
; On parking failure
exten = 7,1,Goto(operator,s,1)
Does this help?
On 8/25/06, J. Oquendo [EMAIL PROTECTED] wrote:
Andrew Latham wrote:
You can have it come back on another line appearance that is set with
different ringtone.
Would you happen to have to have an example context of this? I'm puzzled
by what you mean
--
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http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net
The happiness of society is the end of government.
John Adams
--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
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Re: [asterisk-users] [RESOLVED] One way audion on Sangoma
On 09:48, Fri 25 Aug 06, Dr. Michael J. Chudobiak wrote: No, that is incorrect. From http://wiki.sangoma.com/wanpipe-asterisk-configure: The Wanpipe TDM driver enables HW Echo Cancellation only on channels that have active calls: It waits for zaptel to enable echo cancellation after the call has been established. Therefore, Echo Cancellation option MUST be enabled in /etc/asterisk/zapata.conf. Ah, then it's different from the digium cards. Sorry for the misinformation. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Realtime and hints
If you have to put one of the lines in extensions.conf, then you completely lose all the advantages that realtime gives you. You might as well just put both in extensions.conf now, as any change to the hint, or an addition, deletion etc is going to require a database change which is ok, but also an edit of the file and a subsequent asterisk reload. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Friday, August 25, 2006 7:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints I dont know why it is working but it is. My first line I have in extensions.conf and the second I have in MySql. - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 24, 2006 1:00 PM Subject: RE: [asterisk-users] Realtime and hints I don't see how that helps. If you have a portion of the hint still in extensions.conf, then what use is the database? -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Realtime and hints That's what he was gettin at. Take the second line out, and put the first priority in the database. On Thu, 2006-08-24 at 10:17 -0600, Douglas Garstang wrote: But... you need _both_ in your dialplan. My extensions.conf has: exten = 2944054,hint, SIP/2944054 exten = 2944054,1, Dial(SIP/2944054) ie two lines for the hint. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints Doug I have Exten = 10,hint,SIP/11010 and in mysql I have exten = 10,1,Dial(SIP/11010) - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 21, 2006 3:37 PM Subject: [asterisk-users] Realtime and hints Can realtime be used with hints? How would you get the following into the database given that the priority column is numeric, and that there is no application for the first entry? exten = 2944006,hint,SIP/2944006 exten = 2944006,1,Dial(SIP/2944006) Every time I touch realtime I hit obstacles. How are others getting around this limitation? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New Asterisk Voice Changer 0.4
Hi again! A new version 0.4 of my Voice Changer for Asterisk 1.2 was released 2006-08-23 http://www.lobstertech.com/code/voicechanger/ Yes I finally stopped being lazy and updated it! ^_^ Here are the main features: - Simpler build system, no messy patching! - CDR record handling should work correctly now - Will set the DIALSTATUS variable - Change pitch during conversation with * and # - Voice effect can be applied to peer channel instead with 'p' option The simpler build system is partially due to help from anthm for wrapping my modifications to SoundTouch in to a separate library found here: http://www.lobstertech.com/code/libsoundtouch4c/ Please note that this version is a total rewrite and I haven't had a whole lot of time to valgrind and test it rigorously. If you have ANY problems, even if you figure out how to work around them, please let me know so I can fix it so others don't have the same problem. I usually respond pretty quick to email (in the 5 minutes to one day range) Plans for 0.6: - Change voice pitch via manager api and command line - Open to suggestions Have fun! - Justin Tunney ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [RESOLVED] One way audion on Sangoma
On Friday 25 August 2006 10:46, Michiel van Baak wrote: The Wanpipe TDM driver enables HW Echo Cancellation only on channels that have active calls: It waits for zaptel to enable echo cancellation after the call has been established. Therefore, Echo Cancellation option MUST be enabled in /etc/asterisk/zapata.conf. Ah, then it's different from the digium cards. Sorry for the misinformation. No, it's not. Digium hardware echo cancellation cards also require you to say echocancel=yes in zapata.conf. The zaptel driver recognizes that the card possesses echo cancellation hardware and does not engage the software echo canceller for those channels. To summarize: if you want echo cancellation on Zaptel channels, you must enable it (echocancel=yes, or a number of taps) in zapata.conf. If hardware echo cancellation exists, it is used over software echo cancellation. Note that if hardware echo cancellation hardware is detected, the # of taps is ignored and the hardware uses whatever it has internally. echocancel=no/off in zapata.conf will disable the echo cancellation in Zaptel, whether it is hardware-based or software-based. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] New Asterisk Voice Changer 0.4
Way cool, I've been waiting for an application like this. Anyone out there using it and have any thoughts/feedback for people on the list who haven't tried it. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Justin Tunney Sent: Friday, 25 August 2006 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] New Asterisk Voice Changer 0.4 Hi again! A new version 0.4 of my Voice Changer for Asterisk 1.2 was released 2006-08-23 http://www.lobstertech.com/code/voicechanger/ Yes I finally stopped being lazy and updated it! ^_^ Here are the main features: - Simpler build system, no messy patching! - CDR record handling should work correctly now - Will set the DIALSTATUS variable - Change pitch during conversation with * and # - Voice effect can be applied to peer channel instead with 'p' option The simpler build system is partially due to help from anthm for wrapping my modifications to SoundTouch in to a separate library found here: http://www.lobstertech.com/code/libsoundtouch4c/ Please note that this version is a total rewrite and I haven't had a whole lot of time to valgrind and test it rigorously. If you have ANY problems, even if you figure out how to work around them, please let me know so I can fix it so others don't have the same problem. I usually respond pretty quick to email (in the 5 minutes to one day range) Plans for 0.6: - Change voice pitch via manager api and command line - Open to suggestions Have fun! - Justin Tunney ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] misdn-init.conf card parameter for a monoBRI
Hi, I need to know the card parameter value for a monoBRI (card=1, ???) . Is there anybody who knows it, please? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [RESOLVED] One way audion on Sangoma
On 11:24, Fri 25 Aug 06, Andrew Kohlsmith wrote: Digium hardware echo cancellation cards also require you to say echocancel=yes in zapata.conf. The zaptel driver recognizes that the card possesses echo cancellation hardware and does not engage the software echo canceller for those channels. To summarize: if you want echo cancellation on Zaptel channels, you must enable it (echocancel=yes, or a number of taps) in zapata.conf. If hardware echo cancellation exists, it is used over software echo cancellation. Note that if hardware echo cancellation hardware is detected, the # of taps is ignored and the hardware uses whatever it has internally. echocancel=no/off in zapata.conf will disable the echo cancellation in Zaptel, whether it is hardware-based or software-based. thnx, I'll now go back to reading docs before I say anything stupid again. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 7970 'LoadID incorrect' problem
Hi,
Just trying to setup my 7970 with latest SIP image
(SIP70.8-0-3S)
I referenced the page
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP
And used the following as my
SEPmac.cnf.xml
devicedevicePoolcallManagerGroupmembersmember
priority="0"callManagerportsethernetPhonePort2000/ethernetPhonePort/portsprocessNodeName/processNodeName/callManager/member/members/callManagerGroup/devicePoolversionStamp{Jan
01 2005
00:00:00}/versionStamploadInformationSIP70.8-0-3S/loadInformationaddOnModules/addOnModulesuserLocalenameEnglish_United_States/namelangCodeen/langCode/userLocalenetworkLocale/networkLocaleidleTimeout0/idleTimeoutauthenticationURL/authenticationURLdirectoryURL/directoryURLidleURL/idleURLinformationURL/informationURLmessagesURL/messagesURLproxyServerURL/proxyServerURLservicesURL/servicesURL/device
But I get the 'LoadID incorrect' error
How do I find the correct LoadID?
I simply reset the phone everytime with **#** in
settings
Thanks
Paul
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Re: [asterisk-users] Realtime and hints
Ok. Now I understand. What you can do is put it in a db and whenever you make changes you have asterisk grab the info from the db and put it into a file and reload asterisk. Of course asterisk supporting it would be a lot easier. Maybe you can get some one on the devel. list to do it. I also wondering if they will ever have support for complete real time of contexts. So I dont have to put them in the static files. I have a friend that when he has time will patch asterisk to look up a new table in the db with a list of contexts so that each time u create a new one you can just add it to the DB. Dovid - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 25, 2006 11:09 AM Subject: RE: [asterisk-users] Realtime and hints If you have to put one of the lines in extensions.conf, then you completely lose all the advantages that realtime gives you. You might as well just put both in extensions.conf now, as any change to the hint, or an addition, deletion etc is going to require a database change which is ok, but also an edit of the file and a subsequent asterisk reload. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Friday, August 25, 2006 7:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints I dont know why it is working but it is. My first line I have in extensions.conf and the second I have in MySql. - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 24, 2006 1:00 PM Subject: RE: [asterisk-users] Realtime and hints I don't see how that helps. If you have a portion of the hint still in extensions.conf, then what use is the database? -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Realtime and hints That's what he was gettin at. Take the second line out, and put the first priority in the database. On Thu, 2006-08-24 at 10:17 -0600, Douglas Garstang wrote: But... you need _both_ in your dialplan. My extensions.conf has: exten = 2944054,hint, SIP/2944054 exten = 2944054,1, Dial(SIP/2944054) ie two lines for the hint. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints Doug I have Exten = 10,hint,SIP/11010 and in mysql I have exten = 10,1,Dial(SIP/11010) - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 21, 2006 3:37 PM Subject: [asterisk-users] Realtime and hints Can realtime be used with hints? How would you get the following into the database given that the priority column is numeric, and that there is no application for the first entry? exten = 2944006,hint,SIP/2944006 exten = 2944006,1,Dial(SIP/2944006) Every time I touch realtime I hit obstacles. How are others getting around this limitation? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com --
Re: [asterisk-users] IP phone with 2 ethernet jacks
SNOM is a good phone but dosent have QOS. The polycom does :) - Original Message - From: Guido Hecken [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 25, 2006 4:52 AM Subject: RE: [asterisk-users] IP phone with 2 ethernet jacks We like the SNOM 360 Phones. They have really good features. Guido -Ursprüngliche Nachricht- Von: Mindaugas Kuprys [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 25. August 2006 09:40 An: asterisk-users Betreff: [asterisk-users] IP phone with 2 ethernet jacks Hi, Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wanted Sipura but they don't have such product. Thanks, Mindaugas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] misdn-init.conf card parameter for a monoBRI
On Fri, 2006-08-25 at 17:36 +0200, Giorgio Incantalupo wrote: Hi, I need to know the card parameter value for a monoBRI (card=1, ???) . Is there anybody who knows it, please? First do: # misdn-init scan That should show you what card it is. Next do: # misdn-init config That writes the config to /etc/misdn-init.config. Next load the modules with: # misdn-init start Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Standard for transfer via IAX provider?
Is there any standard way to signal to an IAX provider that I want them to conference in another Asterisk box located elsewhere and then hand off the call to the remote center after a short period of three-way talk? My problem is that I don't want to take a double hit for latency back and forth from the United States. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] IP phone with 2 ethernet jacks
AFAIK snom does support layer 2 and layer 3 QoS. Is there any other QoS? CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Friday, August 25, 2006 12:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IP phone with 2 ethernet jacks SNOM is a good phone but dosent have QOS. The polycom does :) - Original Message - From: Guido Hecken [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 25, 2006 4:52 AM Subject: RE: [asterisk-users] IP phone with 2 ethernet jacks We like the SNOM 360 Phones. They have really good features. Guido -Ursprüngliche Nachricht- Von: Mindaugas Kuprys [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 25. August 2006 09:40 An: asterisk-users Betreff: [asterisk-users] IP phone with 2 ethernet jacks Hi, Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wanted Sipura but they don't have such product. Thanks, Mindaugas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Realtime and hints
We have bounced around various methods of efficient provisioning for months, and unfortunately none of them seem to be a very good solution. In regards to building static files, downloaded from the db: 1. Every time a customer makes a change, we have to download files again, and do a 'reload' again. If several customers are doing it at the same time, this could cause many downloads and reloads together. This doesn't seem like a good solution. We could batch them together, and do them at specified intervals, but then we have to tell customers via our web management interface that they have to wait X number of minutes for their findme/followme call routing changes to become effective. Not very nice. 2. How do you represent the files in the database? Do you store every line as a record? We tried building a tiered hierarchial structure of roles, clusters, hosts, files, contexts and elements for flexibility, but even with MySQL consultant help, it became very complicated. 3. I'm sure I've forgotten some stuff. In regards to using realtime: 1. You can't store BLF in realtime. 2. Realtime doesn't support ex-girlfriend logic. 3. You still need to use the 'include =' statement in the dialplan. This means your still going to have to make edits to the config files anyway from time to time, even with realtime. 4. The data as stored in the db is hard to manipulate for a Web Developer who doesn't know the inner workings of Asterisk. In my mind, provisioning and management are two of Asterisk's BIGGEST challenges. We've been stewing over it for a long time. Doug. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Friday, August 25, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints Ok. Now I understand. What you can do is put it in a db and whenever you make changes you have asterisk grab the info from the db and put it into a file and reload asterisk. Of course asterisk supporting it would be a lot easier. Maybe you can get some one on the devel. list to do it. I also wondering if they will ever have support for complete real time of contexts. So I dont have to put them in the static files. I have a friend that when he has time will patch asterisk to look up a new table in the db with a list of contexts so that each time u create a new one you can just add it to the DB. Dovid - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 25, 2006 11:09 AM Subject: RE: [asterisk-users] Realtime and hints If you have to put one of the lines in extensions.conf, then you completely lose all the advantages that realtime gives you. You might as well just put both in extensions.conf now, as any change to the hint, or an addition, deletion etc is going to require a database change which is ok, but also an edit of the file and a subsequent asterisk reload. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Friday, August 25, 2006 7:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints I dont know why it is working but it is. My first line I have in extensions.conf and the second I have in MySql. - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 24, 2006 1:00 PM Subject: RE: [asterisk-users] Realtime and hints I don't see how that helps. If you have a portion of the hint still in extensions.conf, then what use is the database? -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Realtime and hints That's what he was gettin at. Take the second line out, and put the first priority in the database. On Thu, 2006-08-24 at 10:17 -0600, Douglas Garstang wrote: But... you need _both_ in your dialplan. My extensions.conf has: exten = 2944054,hint, SIP/2944054 exten = 2944054,1, Dial(SIP/2944054) ie two lines for the hint. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints Doug I have Exten = 10,hint,SIP/11010 and in mysql I have exten = 10,1,Dial(SIP/11010) - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List -
Re: [asterisk-users] IP phone with 2 ethernet jacks
Actually Sipura does have a 2 port IP phone but it's branded as the LinksysSPA942 (http://store.voxilla.com/customer/product.php?productid=16204cat=267page=1 ). I don't have any experience with it but my brother uses about a dozen of the single port SPA941s (http://store.voxilla.com/customer/product.php?productid=16199cat=267page=1 ) at his office and says they're pretty good. But at that price I would just go a little higher and get either the Polycom IP501 or IP601, I have the 501 here on my desk and it's great. My only complaint is when configuring it you have to have a lot of patience. On 8/25/06, Mindaugas Kuprys [EMAIL PROTECTED] wrote: Hi,Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wantedSipura but they don't have such product.Thanks,Mindaugas___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] GSM gateway and FXO ATA
Hello
WE can provide you with budget GSM Gateway if you are interested?
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomislav
Par?ina
Sent: Tuesday, August 22, 2006 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] GSM gateway and FXO ATA
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over
Grandstream HT488 ATA. I have plugged gateway to FXO port of ATA. Incoming
calls work the way I call GSM number and then I get DISA to call inside
company. Outgoing call work well when I call VoIP number of ATA which calls
gateway and then I dial number I wish to call over gateway. As I said, it
works fine.
Now I would like to dial ATA_number+number_I_wish_to_call so that I don't
have to dial twice when I'm trying to establish outgoing call from company
thru gateway.
I have tried this but it doesn't work well.
; to dial outside thru GSM gateway
exten = _456.,1,Dial(SIP/${EXTEN},30,tTD(${EXTEN:3}))
exten = _456.,n,Hangup
This is what I see on CLI:
-- Executing Dial(SIP/577-104c, SIP/4560989970434|30|tTD(248)) in
new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup(SIP/577-104c, ) in new stack
== Spawn extension (sip, 4560989970434, 2) exited non-zero on
'SIP/577-104c'
Why asterisk thinks that gateway is busy when it's not?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
___
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[asterisk-users] DNS
Asterisk server is setup in /etc/resolv.conf to query my primary and backup NS. Had an issue with my primary NS and asterisk refused to complete any calls or forward inbound calls to extensions. I had to manually switch it to look at the backup NS first then reboot for it to start working while I fixed the primary. Is this behavior normal or am I missing a step? All hosts, etc are identified by IP. Ver 1.2.10 Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DNS
On 25 aug 2006, at 20.18, Bill Gibbs wrote: Asterisk server is setup in /etc/resolv.conf to query my primary and backup NS. Had an issue with my primary NS and asterisk refused to complete any calls or forward inbound calls to extensions. I had to manually switch it to look at the backup NS first then reboot for it to start working while I fixed the primary. Is this behavior normal or am I missing a step? All hosts, etc are identified by IP. I have had similar issues. To sort of resolve this I had to install a local name-server on the machine that contains the addresses asterisk tries to resolve (changing to using IP-addresses did not fix the issue for me either). I would prefer an option in asterisk that tells it to not resolv more than once on each address. /Ola ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [RESOLUTION] Polycom microbrowser issue Error HTTP 406 withIIS
Title: Message I found this solution from the web and figured I'd share it because it affects all phones getting input from IIS. Map .gif, .jpg, .css etc (in my case I used .xhtml for the Polycom 601) in IIS under your sites: Properties -Virtual directory tab- Configuration - Application configuration - Mappings tab. Make ASP DLL [..\inetsrv\asp.dll] to handle these files. This allows the file with extension XHTML to be passed to the phone and not return a HTTP 406 error (File type not supported by your browser). Hope is helps others. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil MenicoSent: Friday, August 25, 2006 8:51 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Polycom microbrowser issue Error HTTP 406 withIIS Thanks, but we have reasons to want to make it work with IIS. Anyone have a hint of what is the issue? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas GarstangSent: Thursday, August 24, 2006 6:46 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Polycom microbrowser issue Error HTTP 406 withIIS We had a similar problem. Eventuallywe gave up and just used apache. We found that _exactly_ the same content would not work with IIS, but WOULD work with Apache. -Original Message-From: Phil Menico [mailto:[EMAIL PROTECTED]Sent: Thursday, August 24, 2006 3:06 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Polycom microbrowser issue Error HTTP 406 with IIS I have no where else to turn to so if anyone has an answer please send it. I am running sip version 1.6.on a Polycom 601on Asterisk and am unable to get the microbroser to work. The phone returns a 406 error for both idle and services. I can see the file being requested and the subsequent 406 error in the IIS log files. Any ideas on what permissions are needed in IIS or how to format the webpage file? I tried both these 2 files with no luck XHTML file 1: html head /head body Hello phil post /body/html XHTML file 2: ?xml version="1.0" encoding="UTF-8"?html xmlns="http://www.w3.org/1999/xhtml" xml:lang="en" lang="en" head titleVirtual Library/title /head body PHello phil/P /body/html Log info from IIS: 2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET /Polycom/ - 302 0 295 202 0 HTTP/1.1 10.0.1.210:81 Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 - -2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET /Polycom/post.htm - 406 0 4085 242 10 HTTP/1.1 10.0.1.210:81 Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 - http://10.0.1.210:81/Polycom Thank you. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DNS
Ola Lidholm [EMAIL PROTECTED] wrote: To sort of resolve this I had to install a local name-server on the machine that contains the addresses asterisk tries to resolve (changing to using IP-addresses did not fix the issue for me either). I would prefer an option in asterisk that tells it to not resolv more than once on each address. Have you tried setting timeout, attempts and rotate in resolv.conf? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zap channel media volume
I've been fighting with this issue for over a year. There are several threads here talking about it: Digium Zaptel volume issues setting of volume Low volume/audio problems on TDM400 card increase the volume ? There is one thread (Voicemail volume adjustment) that give me hope that this can be fixed that mentions adding |usg(10) to the dial command to increase the gain. I'm still a novice at the inner workings of asterisk so I'm hoping one of the gurus on the list will figure this out eventually. JD Hi all, we do have the following configuration (non-Asterisk PBX) - T1 - ZAP (Asterisk PBX) - ZAP - T1 - (GSM Gateway) - GSM Enduser The call is originated on the (non-Asterisk PBX) - gets send over a T1 connection to the asterisk server (which does least cost routing) - the asterisk server then does send the call over a GSM Gateway into the world... The Problem we do have is - that the Users behind the non-Asterisk PBX are complaining about low volume media if the the calling through the gateway (if the are calling mobiles...). So i have started to raise the rxgain value for the connection between the asterisk box and the GSM Gateway, this does work quite well - but not really perfect. The ringback (not locally generated - does come from the GSM Provider) does get terrible loud - as soon as the callee is connected - the speech is nearly not hearable because it has such a low volume. The ringback is EARLY MEDIA - if i am right - and the speech is normal MEDIA. So, is it possible to set different gains for EARLY MEDIA and normal MEDIA ? Does anyone else have had this problem ? regards, Wolfgang ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Will Asterisk work with Exchange 2007 UM?
I'm faced with the need to create forensic test data for an Exchange 2007 server with unified messaging. Microsoft has a list of tested PBX and IP gateway products that are known to work (below) but I'd prefer to use Asterisk if possible. From everything I've read it appears that since Exchange uses SIP over IP and Asterisk uses SIP over UDP this will not work. I don't have a lot of experience with Asterisk but I was wondering if anyone knows of a plan to allow Asterisk to run SIP over IP or if there are any SIP gateways that will make this conversion. Reading through the Asterisk/Digium documentation and the asterisk-users list archive didn't turn up any clues. I apologize if this topic has already been discussed. Anyone have any ideas? http://www.microsoft.com/technet/prodtechnol/exchange/2007/productevaluation /sysreqs.mspx#pbx Thanks in advance, Brian Lawrence mail2web - Check your email from the web at http://mail2web.com/ . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DNS
Bill Gibbs wrote: Asterisk server is setup in /etc/resolv.conf to query my primary and backup NS. Had an issue with my primary NS and asterisk refused to complete any calls or forward inbound calls to extensions. I had to manually switch it to look at the backup NS first then reboot for it to start working while I fixed the primary. Is this behavior normal or am I missing a step? All hosts, etc are identified by IP. Ver 1.2.10 Most people don't think much about dns, but if your primary dns server responded with anything (including a simple I don't know response), the secondary dns server will not be attempted. So, depending upon exactly what was wrong with your primary, your stated result can be very normal. Regarding asterisk stop responding when no dns server is present, that's been discussed many many times on this list, the latest as of earlier this week. Asterisk code does have a problem, and I'd be reasonably certain part of the problem is the OS underlying dns resolver operates in a blocking mode. In the past, one of the suggested workarounds was to implement a dns caching-only server on the asterisk box. I've not done that and I don't recall hearing anyone's actual experience after doing it. Another suggested workaround is to use IP addresses only in your configs (which is what I've been doing for three years). But, you'll need to make sure nothing in the configs gets interpreted as a dns name. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Idiot questions
As long as they provide some sort of DOA guarantee, I wouldn't worry too much about it. -brandonOn 8/25/06, joea, j4computers [EMAIL PROTECTED] wrote:I actually had a look at one on ebay.What concerned me was the fact that the seller had set it on a carpet for the pictures.I was concerned about static damage.Too late now, tho. I'm still conflicted about it.Sigh.joeaDualcall.com [EMAIL PROTECTED] wrote on 8/25/2006 2:43 AM: Hello, If you can buy a TDM400, good! Support Digium Cheers, Madhawa Nilesh Londhe wrote: I would suggest buying a very low price FXO to begin with which would probably be x100p PCI card at ebay for about $10 +shipping. On 8/24/06, *Adam Collard* [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] wrote: You will need a TDM400 with an FXO module for each line you want. A TDM400 supports up to four lines or analog stations. For two lines, you should get a TDM04B. -Original message- From: joea, j4computers [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] Date: Thu, 24 Aug 2006 14:58:21 -0700 To: asterisk-users@lists.digium.com mailto: asterisk-users@lists.digium.com Subject: [asterisk-users] Idiot questions As a complete newcomer to Asterisk, Digium and PBX, I have several questions. But I'll start with this. To setup a simple system with only a couple of POTS lines, I gather I will need a TDM400 board with FXO and/or FXS modules. So, a TDM400 card will support up to two analog (POTS) lines? joea ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Adam Collard President Digital Telecom of Michigan, Inc. [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] (517) 233-1072 Direct Office (800) 420-3803 x4101 Office (517) 766-5902 Fax This email may be confidential. Any distribution, use or copying of this email or the information it contains by other than an intended recipient is unauthorized. If you received this email in error, please advise me (by return email or otherwise) immediately. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: [EMAIL PROTECTED]AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zap channel media volume
The root cause of the low volume problem is the result of software echo cancellation software, and its need to insert a noticeable loss. If I recall correctly, the wctdm.c driver has a statically defined loss value of something like -6 db that is loaded into the TDM400 chipset at driver load time. Ordinarily, that loss is not all that noticeable. But, if your pstn line is rather lengthy (greater then about 5db worth of loss), the two loss values become very noticeable and marginal to users. There is no known fix or workaround. The low audio becomes even worse when a pstn caller leaves a voicemail and the user calls in via the pstn to retrieve his voicemail. The voicemail gain setting was intended to be sort of a workaround, but its marginal at best. JD Austin wrote: I've been fighting with this issue for over a year. There are several threads here talking about it: Digium Zaptel volume issues setting of volume Low volume/audio problems on TDM400 card increase the volume ? There is one thread (Voicemail volume adjustment) that give me hope that this can be fixed that mentions adding |usg(10) to the dial command to increase the gain. I'm still a novice at the inner workings of asterisk so I'm hoping one of the gurus on the list will figure this out eventually. JD Hi all, we do have the following configuration (non-Asterisk PBX) - T1 - ZAP (Asterisk PBX) - ZAP - T1 - (GSM Gateway) - GSM Enduser The call is originated on the (non-Asterisk PBX) - gets send over a T1 connection to the asterisk server (which does least cost routing) - the asterisk server then does send the call over a GSM Gateway into the world... The Problem we do have is - that the Users behind the non-Asterisk PBX are complaining about low volume media if the the calling through the gateway (if the are calling mobiles...). So i have started to raise the rxgain value for the connection between the asterisk box and the GSM Gateway, this does work quite well - but not really perfect. The ringback (not locally generated - does come from the GSM Provider) does get terrible loud - as soon as the callee is connected - the speech is nearly not hearable because it has such a low volume. The ringback is EARLY MEDIA - if i am right - and the speech is normal MEDIA. So, is it possible to set different gains for EARLY MEDIA and normal MEDIA ? Does anyone else have had this problem ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DNS
Ola Lidholm wrote: On 25 aug 2006, at 20.18, Bill Gibbs wrote: Asterisk server is setup in /etc/resolv.conf to query my primary and backup NS. Had an issue with my primary NS and asterisk refused to complete any calls or forward inbound calls to extensions. I had to manually switch it to look at the backup NS first then reboot for it to start working while I fixed the primary. Is this behavior normal or am I missing a step? All hosts, etc are identified by IP. I have had similar issues. To sort of resolve this I had to install a local name-server on the machine that contains the addresses asterisk tries to resolve (changing to using IP-addresses did not fix the issue for me either). I would prefer an option in asterisk that tells it to not resolv more than once on each address. That won't fix the problem. If that's all you needed, then change your resolver to use /etc/hosts and statically define each item. However, that totally defeats the dynamic purpose of dns. If you configure the dns server (on each asterisk box) to be a caching only server, then it will do the normal dns lookup and cache that translation one time. Asterisk is generally happy with that. However, if the owner of the dns name that you're looking up sets an unreasonable time-to-live for that name, the caching server isn't going to help much on a flaky network. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] read more than 2 digits on festival
Hi everyone Im making some tests with festival and I saw that festiva cant read more than 2 digits. For ejemplo: Festival reads 55 like five five instead of fifty five any idea tnks Regards Javier ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What are my logs telling me here?
G'Day All, I am trying to figure out and correct some of the issues showing up in the messages log but, I am still a newbie and thus, somewhat at a loss, so here goes: NUMBER 1 -- This appears continuously in the log REACHABLE and the UNREACHABLE: Aug 25 15:24:18 NOTICE[1867]: Peer '5103' is now REACHABLE! (418ms / 1000ms)Aug 25 15:24:21 NOTICE[1867]: Peer '5107' is now REACHABLE! (448ms / 1000ms)Aug 25 15:24:23 NOTICE[1867]: Peer '5108' is now REACHABLE! (445ms / 1000ms)Aug 25 15:25:22 NOTICE[1867]: Peer '5103' is now UNREACHABLE! Last qualify: 418Aug 25 15:25:25 NOTICE[1867]: Peer '5107' is now UNREACHABLE! Last qualify: 448Aug 25 15:25:27 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 445Aug 25 15:26:14 NOTICE[1867]: Peer '5103' is now REACHABLE! (448ms / 1000ms)Aug 25 15:26:17 NOTICE[1867]: Peer '5107' is now REACHABLE! (449ms / 1000ms)Aug 25 15:26:19 NOTICE[1867]: Peer '5108' is now REACHABLE! (472ms / 1000ms)Aug 25 15:27:18 NOTICE[1867]: Peer '5103' is now UNREACHABLE! Last qualify: 448Aug 25 15:27:21 NOTICE[1867]: Peer '5107' is now UNREACHABLE! Last qualify: 449Aug 25 15:27:23 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 472 NUMBER 2 -- Why "cause 3" and "Still have a call" Aug 25 11:08:47 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 460Aug 25 11:08:51 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3)Aug 25 11:08:55 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3)Aug 25 11:09:20 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3)Aug 25 11:09:28 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3)Aug 25 11:09:34 NOTICE[1867]: Peer '5103' is now REACHABLE! (457ms / 1000ms)Aug 25 11:09:39 NOTICE[1867]: Peer '5108' is now REACHABLE! (551ms / 1000ms)Aug 25 11:09:56 NOTICE[1867]: Still have a call...Aug 25 11:09:56 NOTICE[1867]: Peer '5001' is now REACHABLE! (83ms / 1000ms)Aug 25 11:40:29 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3)Aug 25 11:40:30 NOTICE[1867]: Peer '5107' is now REACHABLE! (350ms / 1000ms)Aug 25 11:40:31 NOTICE[1867]: Peer '5108' is now REACHABLE! (355ms / 1000ms)Aug 25 11:49:18 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 449Aug 25 11:49:42 NOTICE[1867]: Still have a call...Aug 25 11:49:42 NOTICE[1867]: Peer '5003' is now REACHABLE! (26ms / 1000ms)Aug 25 11:57:04 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3)Aug 25 11:58:14 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) NUMBER 3 -- This is also repeated quite a bit. Aug 24 14:36:30 WARNING[8809]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request)Aug 24 14:36:30 WARNING[8809]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request)Aug 24 14:36:30 WARNING[8809]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Any pointers, documents, help criticisms welcome..Thanks...Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Idiot questions
Ah, I forgot to add that. I asked explicitly about warranty and a couple of other things. To me no answer to the warranty question meant no warranty. For the price it sold for, I' rather get a new one. I think one of the $20.00 ebay OEM's in the ticket for now. joea Brandon Galbraith[EMAIL PROTECTED] Boldly Declared: 8/25/2006 3:09 PM: As long as they provide some sort of DOA guarantee, I wouldn't worry too much about it. -brandon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What are my logs telling me here?
On 8/25/06, Ferguson, Michael [EMAIL PROTECTED] wrote: G'Day All, I am trying to figure out and correct some of the issues showing up in the messages log but, I am still a newbie and thus, somewhat at a loss, so here goes: NUMBER 1 -- This appears continuously in the log REACHABLE and the UNREACHABLE: Aug 25 15:24:18 NOTICE[1867]: Peer '5103' is now REACHABLE! (418ms / 1000ms) Aug 25 15:24:21 NOTICE[1867]: Peer '5107' is now REACHABLE! (448ms / 1000ms) Aug 25 15:24:23 NOTICE[1867]: Peer '5108' is now REACHABLE! (445ms / 1000ms) Aug 25 15:25:22 NOTICE[1867]: Peer '5103' is now UNREACHABLE! Last qualify: 418 Aug 25 15:25:25 NOTICE[1867]: Peer '5107' is now UNREACHABLE! Last qualify: 448 Aug 25 15:25:27 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 445 Aug 25 15:26:14 NOTICE[1867]: Peer '5103' is now REACHABLE! (448ms / 1000ms) Aug 25 15:26:17 NOTICE[1867]: Peer '5107' is now REACHABLE! (449ms / 1000ms) Aug 25 15:26:19 NOTICE[1867]: Peer '5108' is now REACHABLE! (472ms / 1000ms) Aug 25 15:27:18 NOTICE[1867]: Peer '5103' is now UNREACHABLE! Last qualify: 448 Aug 25 15:27:21 NOTICE[1867]: Peer '5107' is now UNREACHABLE! Last qualify: 449 Aug 25 15:27:23 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 472 NUMBER 2 -- Why cause 3 and Still have a call Aug 25 11:08:47 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 460 Aug 25 11:08:51 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) Aug 25 11:08:55 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) Aug 25 11:09:20 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) Aug 25 11:09:28 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) Aug 25 11:09:34 NOTICE[1867]: Peer '5103' is now REACHABLE! (457ms / 1000ms) Aug 25 11:09:39 NOTICE[1867]: Peer '5108' is now REACHABLE! (551ms / 1000ms) Aug 25 11:09:56 NOTICE[1867]: Still have a call... Aug 25 11:09:56 NOTICE[1867]: Peer '5001' is now REACHABLE! (83ms / 1000ms) Aug 25 11:40:29 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) Aug 25 11:40:30 NOTICE[1867]: Peer '5107' is now REACHABLE! (350ms / 1000ms) Aug 25 11:40:31 NOTICE[1867]: Peer '5108' is now REACHABLE! (355ms / 1000ms) Aug 25 11:49:18 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 449 Aug 25 11:49:42 NOTICE[1867]: Still have a call... Aug 25 11:49:42 NOTICE[1867]: Peer '5003' is now REACHABLE! (26ms / 1000ms) Aug 25 11:57:04 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) Aug 25 11:58:14 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) NUMBER 3 -- This is also repeated quite a bit. Aug 24 14:36:30 WARNING[8809]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Aug 24 14:36:30 WARNING[8809]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Aug 24 14:36:30 WARNING[8809]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Any pointers, documents, help criticisms welcome..Thanks...Mike You've probably got qualify= on your peers in sip.conf. So Asterisk is sending out a SIP OPTIONS msg to which it's waiting for the peer's reply. If it doesn't respond, it then marks the peer as unreachable, and you then cannot dial out to the peer because it's state is UNREACHABLE which will cause (status 3) messages. You might consider increasing your qualify= time and see if that corrects your problems. If not, you're going to need to start looking at possible firewall/network interruptions between your Asterisk instance and your devices to see if they are knocking down traffic that might be trying to flow between. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP phone with 2 ethernet jacks
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mindaugas Kuprys wrote: Hi, Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wanted Sipura but they don't have such product. Go for the Linksys SPA-942. It is what the Sipura SPA-841 evolved into. - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRO9aU0tP/KMNOfRbAQKzZQf6AudHZmr905qjhCIZeXwFnLbwtrSzPExz eGTMYAsSNPFz4gzHJQIJ4Mgs0TtGPxgdOYHJ82ml0tDXRB4gkvs4pz1jZbjXyPq6 +Yv8iny6QelgJ5L64z7pKle4zq/0E5mJUhN4iRJAjCmaOosxzrzydkbS1ygydf64 DI2xdZN3TufcE6FGkrIdMbqb2aGdAVf77YNJccnSa36mEV7Wss5onJAMcnbR7rxi rrqVM/WPPiF9XNnbNUMpGtM8Lk1t+9SmSASHMZ9y2LcQ5dmggDM7k/noPYyi92Lz HCVt8EpyHAjTwryJf+3lvEfKJwhe1aFDxnuMNqSL80AV8FZdzTT/Ow== =CLuv -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What are my logs telling me here?
BJ, Thanks much. I do have qualify in my sip.conf (see below) set at 1000. Also, the asterisk box sits on a public ip ( no firewall) but the devices are behind a WatchGuard firewall. Thanks for the pointers. Send me more if you have any. Thanks [5002] type=friend ; either friend (peer+user), peer or user host=dynamic username=5002 secret=5002 context=toll-access canreinvite=no qualify=1000 callerid=5002 disallow=all allow=ulaw allow=alaw [EMAIL PROTECTED] nat=yes dtmfmode=rfc2833 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Friday, August 25, 2006 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What are my logs telling me here? On 8/25/06, Ferguson, Michael [EMAIL PROTECTED] wrote: G'Day All, I am trying to figure out and correct some of the issues showing up in the messages log but, I am still a newbie and thus, somewhat at a loss, so here goes: NUMBER 1 -- This appears continuously in the log REACHABLE and the UNREACHABLE: Aug 25 15:24:18 NOTICE[1867]: Peer '5103' is now REACHABLE! (418ms / 1000ms) Aug 25 15:24:21 NOTICE[1867]: Peer '5107' is now REACHABLE! (448ms / 1000ms) Aug 25 15:24:23 NOTICE[1867]: Peer '5108' is now REACHABLE! (445ms / 1000ms) Aug 25 15:25:22 NOTICE[1867]: Peer '5103' is now UNREACHABLE! Last qualify: 418 Aug 25 15:25:25 NOTICE[1867]: Peer '5107' is now UNREACHABLE! Last qualify: 448 Aug 25 15:25:27 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 445 Aug 25 15:26:14 NOTICE[1867]: Peer '5103' is now REACHABLE! (448ms / 1000ms) Aug 25 15:26:17 NOTICE[1867]: Peer '5107' is now REACHABLE! (449ms / 1000ms) Aug 25 15:26:19 NOTICE[1867]: Peer '5108' is now REACHABLE! (472ms / 1000ms) Aug 25 15:27:18 NOTICE[1867]: Peer '5103' is now UNREACHABLE! Last qualify: 448 Aug 25 15:27:21 NOTICE[1867]: Peer '5107' is now UNREACHABLE! Last qualify: 449 Aug 25 15:27:23 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 472 NUMBER 2 -- Why cause 3 and Still have a call Aug 25 11:08:47 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 460 Aug 25 11:08:51 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) Aug 25 11:08:55 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) Aug 25 11:09:20 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) Aug 25 11:09:28 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) Aug 25 11:09:34 NOTICE[1867]: Peer '5103' is now REACHABLE! (457ms / 1000ms) Aug 25 11:09:39 NOTICE[1867]: Peer '5108' is now REACHABLE! (551ms / 1000ms) Aug 25 11:09:56 NOTICE[1867]: Still have a call... Aug 25 11:09:56 NOTICE[1867]: Peer '5001' is now REACHABLE! (83ms / 1000ms) Aug 25 11:40:29 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) Aug 25 11:40:30 NOTICE[1867]: Peer '5107' is now REACHABLE! (350ms / 1000ms) Aug 25 11:40:31 NOTICE[1867]: Peer '5108' is now REACHABLE! (355ms / 1000ms) Aug 25 11:49:18 NOTICE[1867]: Peer '5108' is now UNREACHABLE! Last qualify: 449 Aug 25 11:49:42 NOTICE[1867]: Still have a call... Aug 25 11:49:42 NOTICE[1867]: Peer '5003' is now REACHABLE! (26ms / 1000ms) Aug 25 11:57:04 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) Aug 25 11:58:14 NOTICE[1867]: Unable to create channel of type 'SIP' (cause 3) NUMBER 3 -- This is also repeated quite a bit. Aug 24 14:36:30 WARNING[8809]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Aug 24 14:36:30 WARNING[8809]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Aug 24 14:36:30 WARNING[8809]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Any pointers, documents, help criticisms welcome..Thanks...Mike You've probably got qualify= on your peers in sip.conf. So Asterisk is sending out a SIP OPTIONS msg to which it's waiting for the peer's reply. If it doesn't respond, it then marks the peer as unreachable, and you then cannot dial out to the peer because it's state is UNREACHABLE which will cause (status 3) messages. You might consider increasing your qualify= time and see if that corrects your problems. If not, you're going to need to start looking at possible firewall/network interruptions between your Asterisk instance and your devices to see if they are knocking down traffic that might be trying to flow between. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by
RE: [asterisk-users] Setting the contact header on outbound INVITE
Not last I heard...I just fought with this yesterday From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael LunsfordSent: Tuesday, August 22, 2006 8:10 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Setting the contact header on outbound INVITE Is there anyway to set the Contact header on outbound INVITEs such as there is for the REGISTER? I would also like to be able to set the Contact header on responses. Thanks, Michael This email may contain confidential information. If you are not the intended recipient, please advise by return email and delete immediately without reading or forwarding to others. -- Cbeyond ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using asterisk to simulate ISDN BRI line
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I am trying to get 2 Asterisk servers to communicate over an ISDN2e line. Each is running Asterisk 1.2.10 + Bristuff driving a HFC-S chipset BRI card. One is configured as bri_net_ptmp, the other as bri_cpe_ptmp. I have built a crossover cable wired as (Rx+) 3 -- 4 (Tx+) (Tx+) 4 -- 3 (Rx+) (Tx-) 5 -- 6 (Rx-) (Rx-) 6 -- 5 (Tx-) However the slave (cpe) Asterisk box always reports that the D-channel is unavailable. While the master (net) box seems happy. Changing to ptp mode (bri_net and bri_cpe signalling) causes both boxes to report a lack of D-channel. Can anybody tell me where I am going wrong? - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRO9sSUtP/KMNOfRbAQJJcAf+Ip/sN3QU8PMMa5bvGKOPTRzzTMw57abA 7MBsD/Hjv2ylIdC/xKD764xCH+A+RPOTJOopKMKyRNHmnuJM9fytODL8xsPtZcPy NhRI93z0causQkPO4ROsfIW2inozBK7NUSYIiCWn1sC+auWHr2YKu2IpFXTNM6mI bwSqAVpm2WRTWijSWqQVckY1bOVFA02RcdhH2MOQb+57iBbALtxvB3sKKmmHToP+ 16cp3ypeYgBTpff3VcuOEfMePulRplRGb4Oq5kzz5+li4s5Ca9Zbgr5BOoMejZ7W Z39Q6QZn5pyjCgvL249OHOc+4xijMzEz8BjHwx4EZAT5Jacl+yF9Zg== =VKyB -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Will Asterisk work with Exchange 2007 UM?
I'm sorry that I don't have an answer for you, but I too am very interested in hearing what anyone has to say about this. On 8/25/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I'm faced with the need to create forensic test data for an Exchange 2007server with unified messaging. Microsoft has a list of tested PBX and IPgateway products that are known to work (below) but I'd prefer to use Asterisk if possible. From everything I've read it appears that sinceExchange uses SIP over IP and Asterisk uses SIP over UDP this will notwork. I don't have a lot of experience with Asterisk but I was wondering if anyone knows of a plan to allow Asterisk to run SIP over IP or if there areany SIP gateways that will make this conversion. Reading through theAsterisk/Digium documentation and the asterisk-users list archive didn't turn up any clues. I apologize if this topic has already been discussed.Anyone have any ideas?http://www.microsoft.com/technet/prodtechnol/exchange/2007/productevaluation /sysreqs.mspx#pbxThanks in advance,Brian Lawrencemail2web - Check your email from the web at http://mail2web.com/ .___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- -m+b ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Will Asterisk work with Exchange 2007 UM?
Interesting, I never knew that Asterisk SIP doesn't run over IP. How in the world is it then running on my IP network? I guess you meant TCP. Anyhow from that link I can see that T1 getway is supported, which means that thru a T1 card Asterisk should be able to work with it. The following links also show that Asterisk will be quite easy to interface with Exchange 2007 UM, although not with SIP over UDP: https://www.microsoft.com/technet/prodtechnol/exchange/E2k7Help/a7cecf59-b93a-413b-bb88-29f2669ef2cf.mspx?mfr=true https://www.microsoft.com/technet/prodtechnol/exchange/E2k7Help/9ed9dc7a-82e4-47da-b341-a64a1c0da8fd.mspx?mfr=true https://www.microsoft.com/technet/prodtechnol/exchange/E2k7Help/76bcdc54-3ec2-408a-bdbe-37826580dd62.mspx?mfr=true These are all around the same type of info that all show that SIP is just an interface, but you can use any other interface to actualy hook into Exchange 2007 UM, including FXO/FXS, and T1. On 8/25/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I'm faced with the need to create forensic test data for an Exchange 2007 server with unified messaging. Microsoft has a list of tested PBX and IP gateway products that are known to work (below) but I'd prefer to use Asterisk if possible. From everything I've read it appears that since Exchange uses SIP over IP and Asterisk uses SIP over UDP this will not work. I don't have a lot of experience with Asterisk but I was wondering if anyone knows of a plan to allow Asterisk to run SIP over IP or if there are any SIP gateways that will make this conversion. Reading through the Asterisk/Digium documentation and the asterisk-users list archive didn't turn up any clues. I apologize if this topic has already been discussed. Anyone have any ideas? http://www.microsoft.com/technet/prodtechnol/exchange/2007/productevaluation /sysreqs.mspx#pbx Thanks in advance, Brian Lawrence mail2web - Check your email from the web at http://mail2web.com/ . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 25, Issue 119
[EMAIL PROTECTED] is believed to have said: I'd expect it to be in Falcom's best interest to support development efforts as it would open the asterisk market to them. Anyone up for creating a bounty-page for this? I would be more than interested! Anyone else? What would be the steps? Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Strange SIP response
Diego, I've encountered this before, let me review a couple of old logs and notes and I'll get back to regarding this. Cheers, SKM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Andrés Asenjo González Sent: Tuesday, August 22, 2006 7:26 PM Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Strange SIP response Rushowr wrote: Have you run SIP DEBUG PEER 192.168.1.60? It may help...tcpdump is also one of my personal favorites Yes, I have used it. The lines are extracted from a sip debug on the CLI. I'm going to paste more lines: Sip read: SIP/2.0 480 Temporarily Unavailable To: sip:[EMAIL PROTECTED]:6198;tag=e4331437 From: 24307022sip:[EMAIL PROTECTED];tag=as288765a2 Via: SIP/2.0/UDP 172.16.1.3:5060;branch=z9hG4bK73129cd1;received=172.16.1.3 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0 7 headers, 0 lines -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.50 Transmitting: ACK sip:[EMAIL PROTECTED]:6198 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.3:5060;branch=z9hG4bK73129cd1 From: 24307022 sip:[EMAIL PROTECTED];tag=as288765a2 To: sip:[EMAIL PROTECTED]:6198;tag=e4331437 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.1.50:6198 -- SIP/EXT25-a454 is circuit-busy == Everyone is busy/congested at this time I have not detected packet losses even. Thanks for your response. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Andres Asenjo G. Sent: Tuesday, August 22, 2006 6:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange SIP response Hi, I am getting the following message on the CLI: -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I really need help with this error. -- MENSAJE ENVIADO CON WMAIL 1.01 UNIVERSIDAD DEL CAUCA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Will Asterisk work with Exchange 2007 UM?
I don't see anything in there thatI'm not doing already (and have been for over a year, with 200 users)with Asterisk 1.0.9, HylaFAX, and Exchange 5.5, with the exception of the text-to-speech stuff which is do-able with Cepstral / Festival and some scripts that hook MAPI on the Exchange server. But who would want your email read back to you over the phone except as an absolute last resort? Dumb. And, I am a *big* Exchange fan. As to Asterisk compatibility, the system requirements note that a SIP-PRImedia gateway is used to communicate with the PBX with the exception of CCM, so it looks like Exchange is exposing it's API to SIP calls with some sort of middleware. Call control on SIP is 5060 TCP and media stream is on UDP, so SIP is SIP regardless if it's CCM or what have you. Looking in Technet, the Exchange 2007 API is (typical of Microsoft) extremely well-documented; API calls are hooked with "cmdlet's". A cmdlet is a .NET class that's wrapped up in a shell executable that "does stuff" just as a Bash shell script does stuff. So my thinking here is that certain SIP calls from the PBXare recieved by this middleware, the corresponding cmdlet is executedon Exchange, and the result returned to the PBX. Same-same the other way around: User "does stuff" in Exchange (such as click on "Play on Phone" icon in Outlook Web Access) and Exchange triggers the middleware (Windows service, probably) to dial the SIP PBX to call the enduser andplay back the voicemail. Specific functionalitysuch as "play back a voicemail" is most likely sent in the SIP controlenvelope, and a media gateway is probably programmed to take the SIP call, call the PBX on a PRI channel, and execute whatever is supposed to happen through DTMF.So it should be possible to get Asterisk to work with Exchange 2007, yes, but it would be a lot of work to reverse-engineer what is actually going on.* Hmm, I should see if the Exchange 2007 DVD is in my MSDN subscription box. Sounds interesting. However, all of this functionality is do-able with Asterisk today and Exchange integration can be as simple as pie, take the "Play on Phone" functionality for example, which can be as simple as modifying Asterisk's voicemail notification email to include a link, when the user clicks on the link, it invokes a web script that drops a .call file to an Asterisk context that calls the user and logs them into voicemail. * I have no idea if this is actually what is going on but I betcha I'm pretty close. This is how Microsoft typically does things. -Original Message-From: Matt Birmingham [mailto:[EMAIL PROTECTED]Sent: Friday, August 25, 2006 3:35 PMTo: asterisk-users@lists.digium.comSubject: Re: [asterisk-users] Will Asterisk work with Exchange 2007 UM? I'm sorry that I don't have an answer for you, but I too am very interested in hearing what anyone has to say about this. On 8/25/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I'm faced with the need to create forensic test data for an Exchange 2007server with unified messaging. Microsoft has a list of tested PBX and IPgateway products that are known to work (below) but I'd prefer to use Asterisk if possible. From everything I've read it appears that sinceExchange uses SIP over IP and Asterisk uses SIP over UDP this will notwork. I don't have a lot of experience with Asterisk but I was wondering if anyone knows of a plan to allow Asterisk to run SIP over IP or if there areany SIP gateways that will make this conversion. Reading through theAsterisk/Digium documentation and the asterisk-users list archive didn't turn up any clues. I apologize if this topic has already been discussed.Anyone have any ideas?http://www.microsoft.com/technet/prodtechnol/exchange/2007/productevaluation /sysreqs.mspx#pbxThanks in advance,Brian Lawrencemail2web - Check your email from the web athttp://mail2web.com/ .___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- -m+b ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys PAP2 Ring Settings
I have a few PAP2-NA that are being mass configured using the instructions on the wiki for the Sipura mass configuration. However, I need to make sure the following settings are in place as follow: Under the Regional Tab, I need the Ring Waveform to be Trapezoid instead of Sinuzoid and the Synchronized Ring to be Yes instead of No. I made an entry in the XML file for Synchronized_Ring which works just fine. However, no matter what I use for the Ring Waveform (Waveform, Ring_Waveform, Regional_Ring_Waveform), the setting is always the default (Sinuzoid). Does anyone know what the XML tag name/ settings need to be for changing the Ring Waveform? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Will Asterisk work with Exchange 2007 UM?
[EMAIL PROTECTED] wrote: I'm faced with the need to create forensic test data for an Exchange 2007 server with unified messaging. Microsoft has a list of tested PBX and IP gateway products that are known to work (below) but I'd prefer to use Asterisk if possible. From everything I've read it appears that since Exchange uses SIP over IP and Asterisk uses SIP over UDP this will not work. I don't have a lot of experience with Asterisk but I was wondering if anyone knows of a plan to allow Asterisk to run SIP over IP or if there are any SIP gateways that will make this conversion. Reading through the Asterisk/Digium documentation and the asterisk-users list archive didn't turn up any clues. I apologize if this topic has already been discussed. Anyone have any ideas? http://www.microsoft.com/technet/prodtechnol/exchange/2007/productevaluation /sysreqs.mspx#pbx Thanks in advance, Brian Lawrence Brian, Your best bet is to use SER or OpenSER. Because either SER supports SIP over both TCP and UDP you can use them as a proxy between Asterisk and Exchange 2007. You cold even run OpenSER/SER on the same machine as Asterisk. If you had some kind of IP level security in place a pretty basic ser.cfg would do the trick, if not you would have to setup some authentication and stuff... -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CentOS4.3 or Debian 3.1r2?
I'm going to be doing a small production system soon using FreePBX, Sangoma A20004D 8FXO card, 3WARE 80062LP SATA card in RAID1. I have been experimenting with CentOS4.3 and Debian Sarge (3.1r2). They both seem to run well and I feel equally comfortable with both of them. Anyone have an recommendations as to which one has the edge? I would really like to use Debian but with Trixbox using CentOS there seems to be more expertise and info online for that one. What to do?! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CentOS4.3 or Debian 3.1r2?
shadowym wrote: I'm going to be doing a small production system soon using FreePBX, Sangoma A20004D 8FXO card, 3WARE 80062LP SATA card in RAID1. I have been experimenting with CentOS4.3 and Debian Sarge (3.1r2). They both seem to run well and I feel equally comfortable with both of them. Anyone have an recommendations as to which one has the edge? I would really like to use Debian but with Trixbox using CentOS there seems to be more expertise and info online for that one. What to do?! IMO: use what you're most comfortable using. I personally prefer the Red Hat way of doing things, but I've used Debian and Ubuntu, and they're both good distros in their own right. The one (already well-known) gotcha with CentOS or RHEL 4 involves the rwlock_t definition in the kernel headers, which affects the zaptel drivers. As for Fedora (or Debian unstable, for that matter), I wouldn't recommend it for a production server unless you desperately need to use some bleeding-edge hardware that isn't supported under more stable distros. Russ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help compiling asterisk-addons on Debian?
Hello All -Running the following:Debian StableAsterisk SVN-branch-1.2-r41069Checked out the following from SVN:asterisk-addons/branches/1.2 When I attempt to compile asterisk-addons I get the following: /usr/src/asterisk-addons$ make/clipasterdev1:/usr/src/asterisk-addons# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c`app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory cdr_addon_mysql.c:38:19: mysql.h: No such file or directorycdr_addon_mysql.c:39:20: errmsg.h: No such file or directoryres_config_mysql.c:53:19: mysql.h: No such file or directoryres_config_mysql.c:54:27: mysql_version.h: No such file or directory res_config_mysql.c:55:20: errmsg.h: No such file or directorymake -C format_mp3 allmake[1]: Entering directory `/usr/src/asterisk-addons/format_mp3'/clipIs this looking for MySQL? I do have MySQL installed and running, a bit confused here anyone have any thouhts? -- --Christopher T Aloi-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help compiling asterisk-addons on Debian?
Do you have the development libraries installed too? I believe on Debian it's something like libmysqlclient From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher AloiSent: Friday, August 25, 2006 8:36 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] Help compiling asterisk-addons on Debian? Hello All -Running the following:Debian StableAsterisk SVN-branch-1.2-r41069Checked out the following from SVN:asterisk-addons/branches/1.2 When I attempt to compile asterisk-addons I get the following: /usr/src/asterisk-addons$ make/clipasterdev1:/usr/src/asterisk-addons# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c`app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory cdr_addon_mysql.c:38:19: mysql.h: No such file or directorycdr_addon_mysql.c:39:20: errmsg.h: No such file or directoryres_config_mysql.c:53:19: mysql.h: No such file or directoryres_config_mysql.c:54:27: mysql_version.h: No such file or directory res_config_mysql.c:55:20: errmsg.h: No such file or directorymake -C format_mp3 allmake[1]: Entering directory `/usr/src/asterisk-addons/format_mp3'/clipIs this looking for MySQL? I do have MySQL installed and running, a bit confused here anyone have any thouhts? -- --Christopher T Aloi-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MySQL CDR
Hi everyone, I finished installing the Xorcom Rapid's Asterisk Packages with amportal (1.10.10), but i wasn't able to find the asterisk-mysql package. Any idea what happened there?, Is there another reposiitory for that package for asterisk 1.0.11. Or could somebody send me the cdr_addon_mysql.so file? Thanks for your responses, -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help compiling asterisk-addons on Debian?
Thanks for the tip!libmysqlclient12-devGot it doneOn 8/25/06, Rushowr [EMAIL PROTECTED] wrote: Do you have the development libraries installed too? I believe on Debian it's something like libmysqlclient From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Christopher AloiSent: Friday, August 25, 2006 8:36 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] Help compiling asterisk-addons on Debian? Hello All -Running the following:Debian StableAsterisk SVN-branch-1.2-r41069Checked out the following from SVN:asterisk-addons/branches/1.2 When I attempt to compile asterisk-addons I get the following: /usr/src/asterisk-addons$ make/clipasterdev1:/usr/src/asterisk-addons# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c`app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory cdr_addon_mysql.c:38:19: mysql.h: No such file or directorycdr_addon_mysql.c:39:20: errmsg.h: No such file or directoryres_config_mysql.c:53:19: mysql.h: No such file or directoryres_config_mysql.c:54:27: mysql_version.h: No such file or directory res_config_mysql.c:55:20: errmsg.h: No such file or directorymake -C format_mp3 allmake[1]: Entering directory `/usr/src/asterisk-addons/format_mp3'/clipIs this looking for MySQL? I do have MySQL installed and running, a bit confused here anyone have any thouhts? -- --Christopher T Aloi-- ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- --Christopher T Aloi-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Real Time Engine - Fails to Connect to MySQL
Debian StableAsterisk SVN-branch-1.2-r41069Hello List.Okay, tonight I dove into Asterisk Real Time.I have the module compiled and installed and have the following setup:MySQL running with a DB created asterisk *how original right? Username 'astuser' password 'foo'I can locally login to MySQL (as user astuser with pw foo) and select the following:mysql show tables;++| Tables_in_asterisk |++ | sip_friends |++1 row in set (0.00 sec)I have a sip_friend 800 setup in the table.My extconfig looks like this:[settings]sippeers = mysql,asterisk,sip_friends My res_config_mysql.conf looks like this:[general]dbhost = 127.0.0.1dbname = asteriskdbuser = astuserdbpass = foodbport = 3306dbsock = /var/run/mysqld/mysqld.sock And here is what I see in the console; i wasn't able to get any more than this out of the debug file either.Anyone have any thoughts? What is error 2013???Thanks :) !Aug 26 01:13:02 ERROR[6256]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1 (err 2013). Check debug for more info.Aug 26 01:13:02 DEBUG[6256]: res_config_mysql.c:652 mysql_reconnect: MySQL RealTime: Cannot Connect (2013): Lost connection to MySQL server during query -- --Christopher T Aloi-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Real Time Engine - Fails to Connect to MySQL
Scratch that email.problem was using 127.0.0.1 - I changed that to localhost and all is well.Can I have the last three hours back now :) On 8/25/06, Christopher Aloi [EMAIL PROTECTED] wrote: Debian StableAsterisk SVN-branch-1.2-r41069Hello List.Okay, tonight I dove into Asterisk Real Time.I have the module compiled and installed and have the following setup:MySQL running with a DB created asterisk *how original right? Username 'astuser' password 'foo'I can locally login to MySQL (as user astuser with pw foo) and select the following:mysql show tables;++| Tables_in_asterisk |++ | sip_friends |++1 row in set (0.00 sec)I have a sip_friend 800 setup in the table.My extconfig looks like this:[settings]sippeers = mysql,asterisk,sip_friends My res_config_mysql.conf looks like this:[general]dbhost = 127.0.0.1dbname = asterisk dbuser = astuserdbpass = foodbport = 3306dbsock = /var/run/mysqld/mysqld.sock And here is what I see in the console; i wasn't able to get any more than this out of the debug file either.Anyone have any thoughts? What is error 2013???Thanks :) !Aug 26 01:13:02 ERROR[6256]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1 (err 2013). Check debug for more info.Aug 26 01:13:02 DEBUG[6256]: res_config_mysql.c:652 mysql_reconnect: MySQL RealTime: Cannot Connect (2013): Lost connection to MySQL server during query -- --Christopher T Aloi-- -- --Christopher T Aloi-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Linksys PAP2 Ring Settings
This works for me on my SPA-3000 ver 3.1.10(GWd). Ring_WaveformTrapezoid/Ring_Waveform Then back to default. Ring_WaveformSinusoid/Ring_Waveform PAP2-NA shouldn't be any different. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Friday, August 25, 2006 6:27 PM To: Non-Commercial Discussion Asterisk Subject: [asterisk-users] Linksys PAP2 Ring Settings I have a few PAP2-NA that are being mass configured using the instructions on the wiki for the Sipura mass configuration. However, I need to make sure the following settings are in place as follow: Under the Regional Tab, I need the Ring Waveform to be Trapezoid instead of Sinuzoid and the Synchronized Ring to be Yes instead of No. I made an entry in the XML file for Synchronized_Ring which works just fine. However, no matter what I use for the Ring Waveform (Waveform, Ring_Waveform, Regional_Ring_Waveform), the setting is always the default (Sinuzoid). Does anyone know what the XML tag name/ settings need to be for changing the Ring Waveform? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SV: E61
On 2006-08-24 06:32:27 -0700, Jon Schøpzinsky [EMAIL PROTECTED] said: I also have this phone, and have stumbled in to the same problem. I just think that it isn't in nokia's interest to change this, as it forces consumers to have some sort of local hardware, that (possibly) only the telecom provider can give them. This forces the users away from using cheaper services. Nokia makes a load from the telecom operators around the world, and are not interested in pissing them off, by letting their users bypass their price structure. Just my 5 cents. This is a bogus non-issue. Your system isn't configured right or the phone is set wrong. I have used my E60 from many locations on NATS outside the local LAN (which is also a NATTED config). I think the it's a conspiracy thing is a red herring. Now, the fact you can't easily get these phone in the US, that's a conspiracy ;~) Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
