Re: [asterisk-users] dial-out call queue
If you want to do this automatically, what you're looking for is a (Predictive) Dialler for Asterisk. There are a few available, both on the commercial and the free side. I'd start by checking out ViciDial /free) and SineDialer (commercial) that are some of the most used ones. Thanks l. On Mon, 22 Oct 2007 22:57:47 +0200, Joao Pereira [EMAIL PROTECTED] wrote: Is it possible to implement a dial-out call queue in Asterisk? My idea is to give Asterisk a list of numbers, and then he makes the calls and delivers the calls to a call queue. Then, the agents will answer the calls. Is this possible? Thanks Regards Joao pereira -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CFP for HITBSecConf2008 - Dubai now open
The CFP for HITBSecConf2008 - Dubai is now open. Our 2008 event is expected to attract over 300 attendees from around the EMEA region and will see keynote speakers Bruce Schneier (Founder and CTO, BT Counterpane) and Jeremiah Grossman (Founder and CTO, White Hat Security). The event is supported and endorsed by the UAE Telecommunications and Regulatory Authority. Being a deep-knowledge technical conference, talks that are more technical or that discuss new and never before seen attack methods are of more interest than a subject that has been covered several times before. Summaries not exceeding 250 words should be submitted (in plain text format) to [EMAIL PROTECTED] for review and possible inclusion in the programme. Submissions are due no later than 1st January 2008. Topics of interest include, but are not limited to the following: # 3G/3.5G/4G Cellular Networks # Apple / OS X vulnerabilities # SS7/Backbone telephony networks # Smart Card Security and Biometric Systems # UMTS, HSDPA, GPRS and CDMA Security # Security of Wimax, WLAN, Bluetooth, GPS and other wireless technology # Analysis of network and security vulnerabilities # Firewall and Intrusion detection technology # Data Recovery and Incident Response # Network Protocol and Analysis # Analysis of malicious code # Applications of cryptographic techniques # Analysis of attacks against networks and machines # File system security PLEASE NOTE: We do not accept product or vendor related pitches. If your talk involves an advertisement for a new product or service your company is offering, please do not submit. Your submission should include: # Name, title, address, email and phone/contact number # Draft of the proposed presentation (in PDF or PowerPoint format), proof of concept for tools and exploits, etc. # Short biography, qualification, occupation, achievement and affiliations (limit 150 words). # Summary or abstract for your presentation (limit 250 words) # Time (max 60 minutes including time for discussion and questions) # Technical requirements (video, internet, wireless, audio, etc.) Each non-resident speaker will receive accommodation for 2 nights/ 3 days. For each non-resident speaker, HITB will cover travel expenses up to USD 1,000.00. HITBSecConf2008 - Dubai http://conference.hitb.org/hitbsecconf2008dubai/ HITBSecConf 2003,2004,2005,2006 Videos are now on Google Video http://www.hackinthebox.org/modules.php?op=modloadname=Newsfile=articlesid=24719mode=threadorder=0thold=0 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
Jared Smith wrote: On Sun, 2007-10-21 at 13:42 +0200, Per Jessen wrote: The SPA-9x1 does support http download, but I don't see how you could change the initial TFTP request to HTTP without manually configuring the phone. Even then I'm not sure it would work - I certainly haven't managed to make any of my SPAs do an auto-config over HTTP. Actually, it's really easy to do. Here's a copy of my spa942.cfg file which I use to point the phone at my web server, as well as upgrade the firmware. Now that I've found my typo, I completely agree :-) I had $MAC instead of $MA, which produces a MAC address in the nn:nn:nn format. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT traversal packet loss measurement
Yitzhak Bar Geva wrote: How can one measure the effect of NAT traversal packet loss? We currently have no solution for NAT traversal for our SIP clients. We've recently completed a setup (see other threads) with a couple of SIP clients behind NAT in their respective home-offices. Took a couple of attempts, but after consulting the list, we have a working setup. What's the simplest method of preventing packet loss due to NAT traversal in a SIP environment? I doubt very much if any loss you're seeing is due to NAT traversal. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729a codecs + Asterisk 1.4.11
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 To know your architecture, use the cmd: cat /proc/cpuinfo After try to start to use the version below (i686): http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-32/codec_g729a_v32_i686.tar.gz Good luck bilal ghayyad a écrit : Dear Marc; I readed your email about the codec G729a and I am now also need to install the codec on my Asterisk. I typed from Asterisk CLI: core show version and I got the following: Asterisk SVN-branch-1.4-r72556 built by root @ localhost.localdomain on a i686 running Linux on 2007-06-30 13:08:08 UTC So I beleive that my processor is i686, correct? But I am not able to know which one to download: The x86-32 or x86-64 ? Can you please advise. Also, the nocona or the opteron versions? Regards Bilal --- Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64/codec_g729a_v32_nocona.tar.gz codec I have registered my license, copied the codec_g729a.so into the /usr/lib/asterisk/modules folder and restarted my asterisk But on the CLI when I type asterisk*CLI show modules like 72 Module Description Use Count codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g729.so Raw G729 data 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_g723.so G.723.1 Simple Timestamp File Format 0 The codec_g729a.so doesn't appear.. Any idea how to solve the problem. Thanks Best Regards, Marc LEURENT __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHHaLSqjpLE0HiOBYRAizdAJ9r8Hm83u/EMDBeaFCseW/XofIIYwCfbKpk xWjhS4+xRj5G9HpQYAEfwhY= =0rDl -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_swift issues
Hi list,
just wanted to answer my own question for general knowledge - turns out
app_swift requires that the voice be set in swift.conf. If the default voice
in swift is not David-8KhZ then it will say no voice is found, even if a
swift voice is installed. Thanks to Earle for helping me with this.
-yair
On 10/22/07, Yair Hakak [EMAIL PROTECTED] wrote:
Hi all,
i'm trying to integrate cepstral and asterisk, and i have a problem i'd
appreciate any help with (i know it's a bit tangential, but i figure this is
the place with the most knowledge of app_swift and asterisk).
I've installed swift from cepstral.com with alison's voice, and it works
fine, from the command line i can do swift hello there -o test.wav and
then i play the wav and it includes the text. All good.
I've also installed app_swift according to the instructions here
http://www.mezzo.net/asterisk/app_swift.html, and show application swift
from the asterisk CLI brings up the application installed.
Now, when i put Swift('This is a test') in the extensions.conf file, i get
the following:
ERROR[3495]: app_cepstral.c:197 cepstral_speak: Failed to set voice.
I have not touched swift.conf (i'm using the defaults), and, i should add
that i have not yet purchased the cepstral voices so that when i run from
the command line it sticks this voice is unlicensed... or something like
that at the beginning of the file, if that makes some kind of difference.
I found the problem referenced here:
http://www.cepstral.com/forum/viewtopic.php?t=56sid=baa6669e9958920393c62510caa47123PHPSESSID=df1bcc629c4b8b37617d2d72c8b0232e
but no solution...
any help will be most appreciated,
thanks,
yair
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Re: [asterisk-users] Best USB Handset and Softphone Combination
- Original Message - From: Erik Anderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 19, 2007 5:53 PM Subject: Re: [asterisk-users] Best USB Handset and Softphone Combination On 10/19/07, Mike Clark [EMAIL PROTECTED] wrote: Do they play well with Vista? Hah - I have no idea. We installed Vista on one laptop here when Dell started shipping it. That lasted about 3 days and 10 support tickets from the user. Then we reverted back to XP. Haven't touched Vista since. -erik Not to start a flame war mut M$ seems to let the cutomer be the Beta tester. IMHO I would stick with XP for now. Seems to be too many issues with Vista (when it comes to VOIP at least and I am sure there are other issues). ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video Conference
snip Thanks for the responce. Have you had any luck at all even with what one might not consider straight forward? I am trying to avoid paying the $1000+ per location needed to purchase something from say Polycom or Tandberg. I would even be willing to do something along the lines of a web app for video and some how tie that together with the voice through Asterisk. Just don't want to look like one of the old dubbed over Japanese movies from when I was a kid (lips move and then a couple seconds later you hear voice). JohnM John, Try contacting [EMAIL PROTECTED] They have some solution there that works with Asterisk. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Internal Data Stream Error
---
Good Morning,
Any help would be grateful to help me understanding
what's wrong...
I have bought 2 g729a licenses to digium and I would
like to have them
works...
My processor is an Intel(R) Xeon(R) CPU
E5310 @ 1.60GHz (4
processors)
so I have downloaded the
http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64
/codec_g729a_v32_nocona.tar.gz
codec
I have registered my license, copied the
codec_g729a.so into the
/usr/lib/asterisk/modules folder and restarted my
asterisk
But on the CLI when I type
asterisk*CLI show modules like 72
Module Description
Use Count
codec_g726.so ITU G.726-32kbps G726
Transcoder
0
format_g729.so Raw G729 data
0
format_g726.so Raw G.726
(16/24/32/40kbps) data
0
format_g723.so G.723.1 Simple
Timestamp File Format
0
The codec_g729a.so doesn't appear..
Any idea how to solve the problem.
Thanks
Best Regards,
Marc LEURENT
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--
Message: 26
Date: Mon, 22 Oct 2007 16:09:39 -0700
From: Ira [EMAIL PROTECTED]
Subject: Re: [asterisk-users] [France CID] Does Zaptel support ETSI
FSK?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii; format=flowed
At 02:18 PM 10/22/2007, you wrote:
;nothing displayed
exten = s,n,Verbose(${CALLERID})
exten = s,n,Verbose(${CALLERIDNAME})
exten = s,n,Verbose(${CALLERIDNUM})
exten = s,n,NoOp(${CALLERID})
exten = s,n,Verbose(${CALLERID})
;CID at last!
exten = s,n,Verbose(${CALLERID(num)})
I'm running Asterisk 1.4. Does someone know why only the last
statement does display the CID number while the others print nothing?
try adding a wait(1) right in the beginning, worked for me.
Ira
--
Message: 27
Date: Mon, 22 Oct 2007 19:15:55 -0400
From: Jason Lixfeld [EMAIL PROTECTED]
Subject: [asterisk-users] Voicemail playback on iPhone
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII; format=flowed
Anyone managed to get this to work? What's the recipe?
--
Message: 28
Date: Tue, 23 Oct 2007 01:35:13 +0200
From: Yitzhak Bar Geva [EMAIL PROTECTED]
Subject: [asterisk-users] NAT traversal packet loss measurement
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
How can one measure the effect of NAT traversal packet loss?
We currently have no solution for NAT traversal for our SIP clients.
There
is no doubt that packets are getting lost. What is not clear is how much
damage this does. On the face of it, everything seems fine. Could this
be
so? Perhaps we're suffering a degradation in quality or our call setup
times
could be improved. How can we measure this?
What's the simplest method of preventing packet loss due to NAT
traversal in
a SIP environment?
Thanks,
Yitzhak
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Message: 29
Date: Mon, 22 Oct 2007 16:38:17 -0700
From: Ron Stephan [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Voicemail playback on iPhone
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Trick question I assume?
It was mind numbingly simple on my iPhone...(though none of the voice
mail worked when London a few weeks ago).
- tap voice mail -
- tap speaker (upper right) until it turns blue (is activate)
- tap the message you want to playback
- use assorted controls to delete - replay etc.
Now...if the question is ... how do you get asterisk voice mail to show
up on an iPhone...I am all ears. Groovy concept - if
anybody has a hack - I'd love to see it.
Elvis
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Lixfeld
Sent: Monday, October 22, 2007 4:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail playback on iPhone
Anyone managed to get this to work? What's the recipe?
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__ NOD32 2607 (20071022) Information
[asterisk-users] G.729 codec between avaya and asterisk
Dear all i have asterisk connected with avaya through E1 back-2-back now when i configure my sip client with g.729 codec then i m not able to put call from asterisk to avaya and when i user g.711 it is working fine so i dont know why i need G.729 on E1 Trunk it is TDM technologies then why my call fail in g.729 case [sip_phone]--[asterisk]-E1[Avaya][analog_phone] Asterisk sip client configure with g.711 alaw/ulaw Avaya phone client configure g.711 alaw/ulaw suggest how do it implement g.729 on this case what change i have to done on both part __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] opensr Vs freeswitch SIP proxy server
Dear all I have plan for 5000 user register on sip server and call to each other according his/her domain ( Relam ) so which one is best for this type of aaplication or stablity to handle thousand of sip reqest i have study of both product but i need input from community end suggest me best one which can easy and stable for my production my reqierment is [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] this all domain on my sip server and place all according his domain not interdomain Regards Satish Patel __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [France CID] Does Zaptel support ETSI FSK?
On Mon, 22 Oct 2007 16:09:39 -0700, Ira [EMAIL PROTECTED] wrote: try adding a wait(1) right in the beginning, worked for me. Thanks but I had this before, and it makes no difference. Jared explained above why CID isn't displayed when using 1.4. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Short format of SIP INVITE - how to change
My Asterisk box send INVITEs in the short form, i.e., f: instead of from, v: instead of via and so on. Is there a way to force asterisk to use full format? thanks Vitaly __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opensr Vs freeswitch SIP proxy server
On 10/23/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have plan for 5000 user register on sip server and call to each other according his/her domain ( Relam ) so which one is best for this type of aaplication or stablity to handle thousand of sip reqest i have study of both product but i need input from community end suggest me best one which can easy and stable for my production my reqierment is [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] this all domain on my sip server and place all according his domain not interdomain Regards Hi for this kind of things OpenSER is the best, even Freeswitch can do the Job, but OpenSER there since long and testing Million users ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 lights not working on subscription
In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. What we found is that even if you get the lights working, they go off after a few days. The BLF lights on the Snom 360s work for me (Asterisk 1.4, Snom 6.5.12 firmware), but I reboot them nightly. I have noticed that the Snom BLFs can stop working if the network is busy for a long period of time (i.e., longer than the re-registration period), like during system-wide backups and yum-upgrades. To avoid this problem, I have a cron job reboot the Snoms nightly after scheduled backups/upgrades. I'm not sure if this is a network congestion issue or a server CPU overload issue, or something else. Anyway, this arrangement does seem to be pretty reliable. To reboot a Snom: http://www.voip-info.org/wiki/view/Asterisk+phone+snom#RebootingaSNOM360320. Hope this helps. - Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opensr Vs freeswitch SIP proxy server
dear ram i have also find many document about freeswitch and openser and i thing openser is best then freeswitch it is also module base as well as handle thousand of sip call and easy to impliment with DB but freeswitch is XML base and i am not familer with XML language thats why from my point of view is it taff task Regards Satish Patel ram [EMAIL PROTECTED] wrote: On 10/23/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have plan for 5000 user register on sip server and call to each other according his/her domain ( Relam ) so which one is best for this type of aaplication or stablity to handle thousand of sip reqest i have study of both product but i need input from community end suggest me best one which can easy and stable for my production my reqierment is [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] this all domain on my sip server and place all according his domain not interdomain Regards Hi for this kind of things OpenSER is the best, even Freeswitch can do the Job, but OpenSER there since long and testing Million users ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Short format of SIP INVITE - how to change
Vitaly wrote: My Asterisk box send INVITEs in the short form, i.e., f: instead of from, v: instead of via and so on. Is there a way to force asterisk to use full format? sip.conf: compactheaders=no afaik it's disabled by default so it was you who enabled it. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force codec order
On 10/22/07, Il Neofita wrote: There is a way to force the order of the codecs in the sip.conf since the allow seams to let know only the accepted codec. I don't know about sip specifically, but from what I recall reading the .conf files use disallow=all and then add codecs one by one, I believe the order in which you add them is the order they are checked in. your server cannot send a stream that the other side cannot decode, so both sides have to agree on a codec. Don't know if that answers your question. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opensr Vs freeswitch SIP proxy server
With all due respect, please try not to make up spellings based on pronunciation. There is no taff task it is tough task. If someone is going to take the time to answer a question, the least we can do is clearly communicate the question. Spellcheck is readily available where needed. My apology for the off-topic response. -- On 10/23/07, satish patel wrote: dear ram i have also find many document about freeswitch and openser and i thing openser is best then freeswitch it is also module base as well as handle thousand of sip call and easy to impliment with DB but freeswitch is XML base and i am not familer with XML language thats why from my point of view is it taff task Regards Satish Patel ram wrote: On 10/23/07, satish patel wrote: Dear all I have plan for 5000 user register on sip server and call to each other according his/her domain ( Relam ) so which one is best for this type of aaplication or stablity to handle thousand of sip reqest i have study of both product but i need input from community end suggest me best one which can easy and stable for my production my reqierment is [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] this all domain on my sip server and place all according his domain not interdomain Regards Hi for this kind of things OpenSER is the best, even Freeswitch can do the Job, but OpenSER there since long and testing Million users ram __ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - Noetel C15K ?
Has anyone had any luck getting an asterisk box to talk to a Nortel C15K softswitch? I've been playing with it for several days and can't seem to pass calls either direction. I know that whike the Nortel says the C15K speaks SIP, it really speaks nortel's implementation of SIP, but I thought I could get it to at least pass simple calls back and forth to an asterisk box. Right now, I can't even get asterisk to register with it. Anyone have any ideas? thanks! register = username:[EMAIL PROTECTED] [nortel] type=friend fromuser=username username=username canreinvite=yes secret=passwd host= 192.168.1.20 disallow=all allow=gsm allow=ulaw allow=alaw dtmfmode=rfc2833 qualify=yes nat=no usereqphone=yes context=from-nortel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Data calls through TDM2400E
Hello. Has anyone been able to successfully make data (dialup modem) calls through a TDM2400E? We're able to make fax and credit card calls fine, but cannot successfully make modem calls using a 56K modem connected to a patch panel connected to an FXS port which then gets bridged to an FXO port connected directly to a phone line. We have 'echocancelwhenbridged=no' set in /etc/asterisk/zapata.conf. We're dialing into a Lucent TNT which drops the call with a cause code of 11 (DCD-Detected-Then-Inactive. The modem detected DCD, but became inactive). The modem call works fine when connected directly to a phone line. Is there anything else that I can do to get this working? Thanks. Stephen Kratzer Network Engineer CTI Networks, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] White noise from TDM2400
Hello. We recently replaced a channel bank in favor of a TDM2400E. After doing so, users began complaining that they could barely hear the remote parties. We increased gain appropriately for each channel which increased the volume of the voices but has also increased the volume of any line noise. It sounds like white noise which goes away when either party talks and returns during silence. Is there any remedy to this? Thanks. Stephen Kratzer Network Engineer CTI Networks, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best USB Handset and Softphone Combination
Hello Steve, I personally use an 'ultra-portable' headset from Logitech, I would recommend this for any end-users that have laptops: http://www.logitech.com/index.cfm/webcam_communications/internet_headsets_phones/devices/223cl=us,en There is also this model for desktops: http://www.logitech.com/index.cfm/webcam_communications/internet_headsets_phones/devices/3622cl=us,en I've been using Counterpath Eyebeam with the USB headset for a year, and I never had a problem. Eyebeam automatically switches it's profile to use the USB mic and headset when it detects thats it's plugged in. We haven't deployed Eyebeam to our clients because of the risk involved with running a phone on an unmanaged desktop. Most of our customers have little desktop management, if any. But I would recommend either Eyebeam or Bria, they have a provisioning system for easier management and the phones can be rebranded. Good luck ;-) Omar On 10/19/07, Steve Totaro [EMAIL PROTECTED] wrote: I have a client that want to try the softphone with USB handsets route to see if hardphones will even be needed. I always push for hardphones (Polycom) so I am not sure about softphones or USB handsets. This is going to be for a 300+ seat call center onsite and many offsite, I plan on using OpenVPN for the offsite machines. Any advice on softphones, handsets, or practical experience with this sort of deployment? It would be very nice if there was a central way of provisioning the phones. All machines are fairly new (newer than two years), they have very strict policies on downloads and streaming. Thanks in advance. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sharing lines with multiple buttons in Cisco 7960?
Has anyone come up with a way of sharing a single SIP registration with two or more line buttons on the Cisco 79x0? This is possible on a Linksys 94x, but I haven't found the magic parameter on the Cisco (assuming there is one). TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX
[EMAIL PROTECTED] wrote: On this machine its the first install, but i get this error 3 month before on an other machine also. I think the debug will bring t much data, cause there is any half second a call try, and its really hard to find this error in the debug file. The only thing i know is if i use a Sangoma card, the problem went. Can i send you a BIG debug file where some of this errors happened? If you can post a debug of a call when it happened and a call when it doesn't happen, that would help the most. Thanks Nico On Fri, 19 Oct 2007, Matthew Fredrickson wrote: [EMAIL PROTECTED] wrote: Hi, I'm running some Asterisk-machines, and on one of them i get this errors in the CLI, but i don't know what that means. Hardware: Digium 4-Port E1 Card with HWEC Intel Pentium D 3 GHz 2 GB RAM SATA Harddisk Supermicro Mainboard Software: latest libpri/zaptel/asterisk of version 1.2 I tried also asterisk version 1.4.x, but there the problem occurs every 10 calls, on asterisk 1.2 its about every 100 calls. Did this recently start, like after you upgraded or is this something that has always been a problem for you since you installed? If it has always been a problem, can you post a `pri debug span x` trace of a call when this happens? That will help to know more about what is going on here. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] White noise from TDM2400
Stephen Kratzer wrote: Hello. We recently replaced a channel bank in favor of a TDM2400E. After doing so, users began complaining that they could barely hear the remote parties. We increased gain appropriately for each channel which increased the volume of the voices but has also increased the volume of any line noise. It sounds like white noise which goes away when either party talks and returns during silence. Is there any remedy to this? Thanks. Do you have the new TDM2400E with the VPMADT032 on it? Also, what version of zaptel are you running? -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force codec order
On 10/23/07, Il Neofita [EMAIL PROTECTED] wrote: There is a way to force the order of the codecs in the sip.conf since the allow seams to let know only the accepted codec. Hi yes you can do, at client side and as well as Asterisk side. disallow=all allow=first codec allow=second one so on ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opensr Vs freeswitch SIP proxy server
On 10/23/07, satish patel [EMAIL PROTECTED] wrote: dear ram i have also find many document about freeswitch and openser and i thing openser is best then freeswitch it is also module base as well as handle thousand of sip call and easy to impliment with DB but freeswitch is XML base and i am not familer with XML language thats why from my point of view is it taff task Hi i recomend to spend some time and read the documents, and see what is the best to suite your need and find out your own capabilities to deploy the solution. if you feel the task can not achive by you. then opt some cosultant or use some commercial software available to do the best. ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk sends packets on 8004/udp
For the life of me can't figure out why the Asterisk server generates an enormous quantity of outgoing packets on port 8004/udp. They seem to have no effect whether they are blocked by the firewall or not. We're running SIP. Everything appears to be OK (except a large number of ChanSpy write buffer overflow messages, which I also don't understand). How can I discover whether these packets are desirable or not? What should I do with them? I have found no documentation on port 8004. Thanks in advance, Yitzhak Bar Geva ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk sends packets on 8004/udp
For the life of me can't figure out why the Asterisk server generates an enormous quantity of outgoing packets on port 8004/udp. They seem to have no effect whether they are blocked by the firewall or not. We're running SIP. Everything appears to be OK (except a large number of ChanSpy write buffer overflow messages, which I also don't understand). How can I discover whether these packets are desirable or not? What should I do with them? I have found no documentation on port 8004. Thanks in advance, Yitzhak Bar Geva ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sharing lines with multiple buttons in Cisco 7960?
I am not sure if this is what you are looking for but I have this setup on a Cisco 7971. I just duplicated all of the line button=1 entries as line button2 and I can have 2 calls going simultaneously, put each on hold, etc and only one sip peer shows in FreePBX. I'm using Trixbox 2.3.0.2 with Asterisk 1.4.10.1 and FreePBX 2.3.1.0. Pasted appropriate config lines below. Regards Glenn sipLines line button=1 featureID9/featureID featureLabel1187/featureLabel proxy192.168.1.50/proxy port5060/port name1187/name displayName1187/displayName autoAnswer autoAnswerEnabled2/autoAnswerEnabled /autoAnswer callWaiting3/callWaiting authName1187/authName authPassword1187/authPassword sharedLinefalse/sharedLine messageWaitingLampPolicy3/messageWaitingLampPolicy messagesNumber*97/messagesNumber ringSettingIdle4/ringSettingIdle ringSettingActive5/ringSettingActive contact1187/contact forwardCallInfoDisplay callerNametrue/callerName callerNumberfalse/callerNumber redirectedNumberfalse/redirectedNumber dialedNumbertrue/dialedNumber /forwardCallInfoDisplay /line line button=2 featureID9/featureID featureLabel1187/featureLabel proxy192.168.1.50/proxy port5060/port name1187/name displayName1187/displayName autoAnswer autoAnswerEnabled2/autoAnswerEnabled /autoAnswer callWaiting3/callWaiting authName1187/authName authPassword1187/authPassword sharedLinefalse/sharedLine messageWaitingLampPolicy3/messageWaitingLampPolicy messagesNumber*97/messagesNumber ringSettingIdle4/ringSettingIdle ringSettingActive5/ringSettingActive contact1187/contact forwardCallInfoDisplay callerNametrue/callerName callerNumberfalse/callerNumber redirectedNumberfalse/redirectedNumber dialedNumbertrue/dialedNumber /forwardCallInfoDisplay /line -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Komito Sent: Tuesday, October 23, 2007 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Sharing lines with multiple buttons in Cisco 7960? Has anyone come up with a way of sharing a single SIP registration with two or more line buttons on the Cisco 79x0? This is possible on a Linksys 94x, but I haven't found the magic parameter on the Cisco (assuming there is one). TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk sends packets on 8004/udp
On 10/23/07, Yitzhak Bar Geva [EMAIL PROTECTED] wrote: For the life of me can't figure out why the Asterisk server generates an enormous quantity of outgoing packets on port 8004/udp. They seem to have no effect whether they are blocked by the firewall or not. We're running SIP. Everything appears to be OK (except a large number of ChanSpy write buffer overflow messages, which I also don't understand). How can I discover whether these packets are desirable or not? What should I do with them? I have found no documentation on port 8004. Thanks in advance, Yitzhak Bar Geva The only normal service on port 8004 I've found documentation on is Shoutcast (icecast), and people do sometimes configure Asterisk to use shoutcast streams for music on hold. Other than that, most of the rest of the info on 8004 (on Google anyways) is about a remotely accessible exploit again certain Symantec anti-virus engines. So if it isn't a shoutcasted audio stream you've forgotten about, you might want to check if your server has been compromised and is being used to scan the rest of the net for vulnerable Symantec software or something like that. -- Brandon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] text management
Hi, I know that Asterisk doesn't support Instant Messaging, but I'm trying to use the AGI function RECEIVE TEXT to implement a kind of IM service. I have a sip softphone that tries to send a message to an active channel and the AGI script that expect to receive the text through the STDIN. Two problems arise: First: How can I say to asterisk to get the message? (I see on CLI console that the message arrives to asterisk but it drops it) Second: how can I put this message in STDIN to let the AGI read it? Has anybody used this feature? Can someone give an example of how to use it? Any asuggestion would be appreciated.. thank you. Silvia ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk under VMWare
Anyone had any experience with an Asterisk server as a VMWare virtual machine? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Data calls through TDM2400E
Stephen Kratzer wrote: Hello. Has anyone been able to successfully make data (dialup modem) calls through a TDM2400E? We're able to make fax and credit card calls fine, but cannot successfully make modem calls using a 56K modem connected to a patch panel connected to an FXS port which then gets bridged to an FXO port connected directly to a phone line. We have 'echocancelwhenbridged=no' set in /etc/asterisk/zapata.conf. We're dialing into a Lucent TNT which drops the call with a cause code of 11 (DCD-Detected-Then-Inactive. The modem detected DCD, but became inactive). The modem call works fine when connected directly to a phone line. Is there anything else that I can do to get this working? Thanks. I dunno about TDM2400E, but perhaps you might get a connection if you could slow the modem down by using the appropriate extra settings for the modem (IE +ms=v34 or -v90=0 etc) to force a non 56K/v90 connection? -Troy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Data calls through TDM2400E
Stephen Kratzer wrote: cannot successfully make modem calls using a 56K modem connected to a patch panel connected to an FXS port which then gets bridged to an FXO port connected directly to a phone line. We have 'echocancelwhenbridged=no' set in /etc/asterisk/zapata.conf. We're dialing into a Lucent TNT which drops You cannot make 56Kbps modem calls when there is more than one digital/analog transition in the call path. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk under VMWare
On 18:51, Tue 23 Oct 07, WipeOut wrote: Anyone had any experience with an Asterisk server as a VMWare virtual machine? We are running multiple sites as a VMWare virtual machine. All of them are voip only, so I have no idea how it works with T1/E1/POTS interface cards, but as a pure voip setup it works great. -- 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 http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk under VMWare
Michiel van Baak wrote: On 18:51, Tue 23 Oct 07, WipeOut wrote: Anyone had any experience with an Asterisk server as a VMWare virtual machine? We are running multiple sites as a VMWare virtual machine. All of them are voip only, so I have no idea how it works with T1/E1/POTS interface cards, but as a pure voip setup it works great. I've understood that VMWare virtual machines cannot access physical hardware in the host machine directly, only the emulated hardware VMWare provides. As such, no T1/E1/POTS interfaces that connect via PCI. I'm not sure about the suitability of this route, but I would assume that any T1/E1/POTS USB devices out there would work, if they exist, because VMWare *does* allow the host's USB devices to connect to the guest.. Wasn't there talk on the list lately about a USB DS3 adapter? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk under VMWare
Michiel van Baak wrote: On 18:51, Tue 23 Oct 07, WipeOut wrote: Anyone had any experience with an Asterisk server as a VMWare virtual machine? We are running multiple sites as a VMWare virtual machine. All of them are voip only, so I have no idea how it works with T1/E1/POTS interface cards, but as a pure voip setup it works great. Our testing has yielded pretty good results. We had 10 simultaneous calls with ulaw and quality was very good. We are pure VOIP also. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk under VMWare
Quoting WipeOut [EMAIL PROTECTED]: Anyone had any experience with an Asterisk server as a VMWare virtual machine? I was trying to run it under XEN and got into trouble so in all my searches, the conclusion was that running it under VMWare didn't work because of the faulty timer in VMWare... My problem with XEN was due to the fact that I needed access to a PRI card which I never managed to do (didn't try hard enough?). But my Asterisk at home works just perfect under XEN. I don't need hardware access there... XEN solves the timing issue different from VMWare so... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with making calls
[EMAIL PROTECTED] wrote: ZAP/G1 will dial starting from channel 1. ZAP/R1 will dial starting from the last channel of the group. Actually, ZAP/g1 mean start with first channel and work up. ZAP/G1 mean start with last channel and work down. 'r' and 'R' operate in similar directions, but they also remember where they left off last time -- Round-Robin. http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels under Dialing a Group ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is GoVarion a fraud ???
Hi, Some days ago I spent about US$700,00 in a Tormenta III board in www.govarion.com. I used credit card. I didn't receive any answer for my emails and there is no telephone number to contact them.. Now, I'd like to cancel this order, because I couldn´t wait so long, and my credit card was billed. Is www.govarion.com a fraud Does anybody know something about them ?? Thanks. Luis Antonio Prata Barbosa ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk under VMWare
Mike Clark wrote: Michiel van Baak wrote: On 18:51, Tue 23 Oct 07, WipeOut wrote: Anyone had any experience with an Asterisk server as a VMWare virtual machine? We are running multiple sites as a VMWare virtual machine. All of them are voip only, so I have no idea how it works with T1/E1/POTS interface cards, but as a pure voip setup it works great. Our testing has yielded pretty good results. We had 10 simultaneous calls with ulaw and quality was very good. We are pure VOIP also. Excellent.. Thats very positive.. Now to find a UK based IAX trunk outbound call provider.. :) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] register = to let Asterisk register to another softswitch via SIP
Dear Alex; Thanks alot for your nice help. This is if I need to let Asterisk register with another softswitch (so I used register =), what if I need asterisk to send call for the softswitch without register to it (directly)? If I removed the register = then how it will distiguish the IP address in the host at the [sip_trunk] is the IP address of the softswitch that need to register with it and not the IP address of the original caller sip endpoint? Your help is highly appreciated. Regards Bilal The same way you do it with IAX2, pretty much. http://www.voip-info.org/wiki-Asterisk+config+sip.conf On Fri, 19 Oct 2007, bilal ghayyad wrote: Hi All; Alot of softswitches or PBX's does not accept to manipulate any SIP call without being registered firstly. So that means, I need asterisk to register firstly then I can route my calls to that SIP trunk. In IAX2, we use the register = , so what shall we do in Asterisk? And how its format will be (if we will use register)? Or what is the solution? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] register = to let Asterisk register to another softswitch via SIP
Bilal, On Tue, 23 Oct 2007, bilal ghayyad wrote: This is if I need to let Asterisk register with another softswitch (so I used register =), what if I need asterisk to send call for the softswitch without register to it (directly)? If I removed the register = then how it will distiguish the IP address in the host at the [sip_trunk] is the IP address of the softswitch that need to register with it and not the IP address of the original caller sip endpoint? Unless I am missing something here, I suppose the answer is that Asterisk can distinguish the IP endpoints because they are ... distinct. Here is the essence of the situation: In Asterisk it is possible to peer with an endpoint with and without registrations. Registrations are mostly intended for dynamic endpoints whose IP address can potentially change, such as end-user phones off of broadband connections, or other clients whose IP address is not desirable to track or cannot be trusted. The other type of peer is a 'trusted' trunk tied to a particular IP endpoint on the far end. The trust can be done only by IP address, or by IP address + SIP UDP port. This type of peer is typically used when doing SIP handoff from origination and termination carriers on any kind of large-scale, or in other intra-industrial and/or internal and/or intra-platform SIP connections where it is not desirable to position one endpoint of the SIP trunk as a UAC (client) registering against a UAS (server) per se, as such, in the respect that one challenges the other for authentication credentials. So, what I would do is build a trusted trunk (type=peer, insecure=very) to the softswitch that has a static IP (host=) endpoint defined. Then, Asterisk can accept registrations from your users. Where to route the call is determined entirely in the dial plan (extensions.conf), where you can send calls to particular SIP peers. So, for example, here is a regular user defined in sip.conf: [Alex_Evariste_2] type=friend host=dynamic canreinvite=no username=Alex_Evariste_2 secret=xx nat=yes allow=ulaw qualify=yes [EMAIL PROTECTED] context=default-user-dial And here is a dedicated trunk to a provider: [my_sip_provider] host=xxx.yyy.zzz.www insecure=very type=peer qualify=no canreinvite=no dtmfmode=rfc2833 Then, your dial plan for a user can be set up like this, for example, in extensions.conf: [default-user-dial] ; Any North American ten-digit number. exten = _NX,1,Dial(SIP/[EMAIL PROTECTED]) In our case, we actually register with our SIP origination provider, so we have this IP trunk: [junction_networks] fromdomain=jnctn.net host=sip.jnctn.net port=5060 insecure=very username=this_user secret=this_password type=peer qualify=no canreinvite=no dtmfmode=rfc2833 But in addition, in the [general] context at the top of sip.conf, we have: register = our_user:[EMAIL PROTECTED] As you can see, one type of registration requirement does not interfere with another. Hope this helps. If it doesn't, please let me know if I misunderstood something. Cheers, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] register = to let Asterisk register to another softswitch via SIP
P.S. On Tue, 23 Oct 2007, Alex Balashov wrote: [junction_networks] fromdomain=jnctn.net host=sip.jnctn.net port=5060 insecure=very username=this_user-- secret=this_password -- type=peer qualify=no canreinvite=no dtmfmode=rfc2833 The 'username' and 'secret' there are actually not required unless we were to challenge Junction in other direction, which would be impossible with a trunk defined as 'insecure=very' anyway. But since we receive no calls from them, it is completely unnecessary in every respect. Not sure why we have it. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is GoVarion a fraud ???
Luis Antonio Prata Barbosa wrote: Hi, Some days ago I spent about US$700,00 in a Tormenta III board in www.govarion.com http://www.govarion.com. I used credit card. I didn't receive any answer for my emails and there is no telephone number to contact them.. Now, I'd like to cancel this order, because I couldn´t wait so long, and my credit card was billed. Is www.govarion.com http://www.govarion.com a fraud Does anybody know something about them ?? Thanks. Luis Antonio Prata Barbosa Hi, I don't know if it is a scam or not, but I wouldn't pay $49 for a clone x100p FXO card. Alan -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is GoVarion a fraud ???
Luis Antonio Prata Barbosa wrote: Hi, Some days ago I spent about US$700,00 in a Tormenta III board in www.govarion.com http://www.govarion.com. I used credit card. I didn't receive any answer for my emails and there is no telephone number to contact them.. Now, I'd like to cancel this order, because I couldn´t wait so long, and my credit card was billed. Is www.govarion.com http://www.govarion.com a fraud Does anybody know something about them ?? Thanks. Luis Antonio Prata Barbosa It's a bit unusual for a web site that sells things to not have ANY address anywhere on the site. And their Ts Cs and privacy policy are the shortest I've ever seen. Sorry but it does look a bit dubious to say the least... Do you even know which country they operate in? I'd contact your credit card company straight away and let them investigate... Alan -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is GoVarion a fraud ???
This is a problem with smaller companies - when there are only a couple people running things, and something happens, all the customers panic, and even if it was nothing, it could ruin a companies reputation. I am not saying anything one way or the other about govarion since I have had no direct contact with them, but I can say this - one of our customers was forced to evacuate this week due to the fires in san diego - how many of his customers are going to panic when the phones go unanswered ? His location there is simply to answer the phones, no products of the company or real processing happens there, but it takes time to get a number rerouted somewhere else and continue. I realize as a customer its sometimes trying on the patience, but the reality is sometimes things happen beyond a company's control and it has nothing to do with fraud of any sort, or the ability to rely on the company in the future. Quoting Alan Lord [EMAIL PROTECTED]: Luis Antonio Prata Barbosa wrote: Hi, Some days ago I spent about US$700,00 in a Tormenta III board in www.govarion.com http://www.govarion.com. I used credit card. I didn't receive any answer for my emails and there is no telephone number to contact them.. Now, I'd like to cancel this order, because I couldn´t wait so long, and my credit card was billed. Is www.govarion.com http://www.govarion.com a fraud Does anybody know something about them ?? Thanks. Luis Antonio Prata Barbosa Hi, I don't know if it is a scam or not, but I wouldn't pay $49 for a clone x100p FXO card. Alan -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is GoVarion a fraud ???
On 10/23/07, Alan Lord [EMAIL PROTECTED] wrote: Luis Antonio Prata Barbosa wrote: Hi, Some days ago I spent about US$700,00 in a Tormenta III board in www.govarion.com http://www.govarion.com. I used credit card. I didn't receive any answer for my emails and there is no telephone number to contact them.. Now, I'd like to cancel this order, because I couldn´t wait so long, and my credit card was billed. Is www.govarion.com http://www.govarion.com a fraud Does anybody know something about them ?? Not sure about fraud, but I did find this: http://threebit.net/mail-archive/asterisk-users/msg37367.html Somebody complaining about them in 2006. It looks like they exist, they're just very very slow. Good luck! - Gonzalo ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Phone and bitmaps
I've been trying to get the polycom 550 phones to show a idle display bitmap but have not been successful. Anybody have any experience with this? The manual gives instructions (http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soundstation_ip_administrators_guide_v2_2.pdf) but they do not seam to work. So far i've done the following in my sip.conf bitmaps IP_500 . bitmap.IP_500.67.name=mylogo/ /bitmaps indicators ind.idleDisplay.enabled=1 ind.idleDisplay.mode= Animations IP_500 IDLE_DISPLAY ind.anim.IP_500.29.frame.1.bitmap=mylogo ind.anim.IP_500.29.frame.1.duration=0/ /IP_500 /Animations Anybody know where i'm going wrong, watched the ftp logs and i dont see the phone downloading the mylogo.bmp either. Nothing in the -app.log either about it. ~Shaun ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk under VMWare
On Tue, 2007-10-23 at 20:58 +0200, Turbo Fredriksson wrote: [snip] My problem with XEN was due to the fact that I needed access to a PRI card which I never managed to do (didn't try hard enough?). There is a Xen page called something like cool configurations. It has information how you can configure access to a PCI card. Iirc it is even possible to assign one PCI slot/card to one virtual client and another PCI slot to another virtual client. Thanks to CentOS' Andreas Rogge for finding that info for me at the T-DOSE conference. Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Phone and bitmaps
You aren't including the file extension when referencing the graphic name, are you? If so, that would be the problem. You might also want to try loading the parameters to the fields for the 650 also. Just a couple of ideas. Bryan M. Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Shaun R. [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, October 23, 2007 4:52:26 PM (GMT-0500) America/New_York Subject: [asterisk-users] Polycom Phone and bitmaps I've been trying to get the polycom 550 phones to show a idle display bitmap but have not been successful. Anybody have any experience with this? The manual gives instructions (http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soundstation_ip_administrators_guide_v2_2.pdf) but they do not seam to work. So far i've done the following in my sip.conf bitmaps IP_500 . bitmap.IP_500.67.name=mylogo/ /bitmaps indicators ind.idleDisplay.enabled=1 ind.idleDisplay.mode= Animations IP_500 IDLE_DISPLAY ind.anim.IP_500.29.frame.1.bitmap=mylogo ind.anim.IP_500.29.frame.1.duration=0/ /IP_500 /Animations Anybody know where i'm going wrong, watched the ftp logs and i dont see the phone downloading the mylogo.bmp either. Nothing in the -app.log either about it. ~Shaun ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk under VMWare
As a pure SIP solution, we have switched as many as 120 call paths through a VM on a lightly populated host. Bryan M. Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: WipeOut [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 23, 2007 1:51:23 PM (GMT-0500) America/New_York Subject: [asterisk-users] Asterisk under VMWare Anyone had any experience with an Asterisk server as a VMWare virtual machine? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video Conference
Good to hear someone is using WiredRed. I suggested that as an alternative several times on this list but to be honest I'm still astounded that there isn't an asterisk alternative. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Patrick Davis Sent: Tuesday, 23 October 2007 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Video Conference we use WiredRed with some success. I've tried dimdim and it was ok, but not as good as WiredRed. with WiredRed its still going to cost $3k plus per year. it'll do the video, voice and desktop sharing. decent video and audio. Patrick Patrick Davis Study Abroad Canada P.O. Box 3231 51 Univeristy Ave. Charlottetown, PE Canada C1A 7N9 Tel: 902-628-2379 Fax: 902-892-1198 www.studyincanada.ca [EMAIL PROTECTED] On 23-Oct-07, at 6:18 AM, Dovid B wrote: snip Thanks for the responce. Have you had any luck at all even with what one might not consider straight forward? I am trying to avoid paying the $1000+ per location needed to purchase something from say Polycom or Tandberg. I would even be willing to do something along the lines of a web app for video and some how tie that together with the voice through Asterisk. Just don't want to look like one of the old dubbed over Japanese movies from when I was a kid (lips move and then a couple seconds later you hear voice). JohnM John, Try contacting [EMAIL PROTECTED] They have some solution there that works with Asterisk. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk H323 Config
The same as you would treat any other channel. Specify the default context in ooh323.conf (my personal favorite h323 driver) and in extensions.conf under that context set where you want the call to go. - Original Message - From: Arun Kumar To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, October 21, 2007 9:20 AM Subject: [asterisk-users] Asterisk H323 Config Hi Need help on this setup: Incoming DID in H323 Asterisk Server -- SIP Phone please tell me to achieve this above setup what needs to be done in Asterisk. thanks Arun -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tech prefix
You have {EXTEN}shouldnt it be ${EXTEN} (or is this AEL) ?
- Original Message -
From: Jon Weisman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 22, 2007 6:36 PM
Subject: Re: [asterisk-users] tech prefix
no that didnt work.
- Original Message -
From: Philipp Kempgen [EMAIL PROTECTED]
To: Asterisk Users asterisk-users@lists.digium.com
Sent: Tuesday, October 16, 2007 3:09 PM
Subject: Re: [asterisk-users] tech prefix
Jon Weisman wrote:
How can I add a prefix to an outbound call?
_X. = {
Dial(tech/123{EXTEN});
}
?
Regards,
Philipp Kempgen
--
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Re: [asterisk-users] Video Conference
we use WiredRed with some success. I've tried dimdim and it was ok, but not as good as WiredRed. with WiredRed its still going to cost $3k plus per year. it'll do the video, voice and desktop sharing. decent video and audio. Patrick Patrick Davis Study Abroad Canada P.O. Box 3231 51 Univeristy Ave. Charlottetown, PE Canada C1A 7N9 Tel: 902-628-2379 Fax: 902-892-1198 www.studyincanada.ca [EMAIL PROTECTED] On 23-Oct-07, at 6:18 AM, Dovid B wrote: snip Thanks for the responce. Have you had any luck at all even with what one might not consider straight forward? I am trying to avoid paying the $1000+ per location needed to purchase something from say Polycom or Tandberg. I would even be willing to do something along the lines of a web app for video and some how tie that together with the voice through Asterisk. Just don't want to look like one of the old dubbed over Japanese movies from when I was a kid (lips move and then a couple seconds later you hear voice). JohnM John, Try contacting [EMAIL PROTECTED] They have some solution there that works with Asterisk. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Phone and bitmaps
Shaun R. wrote: IP_500 IDLE_DISPLAY ind.anim.IP_500.29.frame.1.bitmap=mylogo ind.anim.IP_500.29.frame.1.duration=0/ /IP_500 Three things, 1). Make sure the logo is in the root ftp directory for that profile, I have a ftp user called polycom and I had to make sure that it was in a directory that the profile had access to. (i.e. /home/polycom) 2). The above config setting is for the IP500, if you have an entry for IP_550, I think you need to use that section. 3). Don't include the file extension. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 codec between avaya and asterisk
Am Dienstag, den 23.10.2007, 02:56 -0700 schrieb satish patel: Dear all i have asterisk connected with avaya through E1 back-2-back now when i configure my sip client with g.729 codec then i m not able to put call from asterisk to avaya and when i user g.711 it is working fine so i dont know why i need G.729 on E1 Trunk it is TDM technologies then why my call fail in g.729 case Hi Satish, Neither do I know why you _need_ G.729. Are there any specific reasons why you do not want to use G711 in the sip client, which is working fine? (Nota bene: there are some more codecs supported by asterisk, some of which may be also supported by your sip phone) Your E1 trunk obviously is G711-only - this is to be expected, because the G711 wave samples are those which go over the wire (as time-division multiplexed bitstream). Together with the information from http://www.voip-info.org/wiki-Asterisk+G.729+Licensing namely ** G.729 requires a license per channel unless it is used ** in pass-thru mode. which exactly matches your setup (by the way that was the first google match for g729 asterisk) we can guess that you did not buy the license which would be necessary for asterisk to transcode G729/G711. [sip_phone]--[asterisk]-E1[Avaya][analog_phone] Asterisk sip client configure with g.711 alaw/ulaw Avaya phone client configure g.711 alaw/ulaw suggest how do it implement g.729 on this case what change i have to done on both part Avaya / E1 stays as is, sip client stays as is, your credit card data is transferred to digium, and their license goes into the appropriate file on your asterisk machine hard drive. Others may have real world experience with those steps, but that is what I read on this mailing list. YMMV, Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 lights not working on subscription
The Linksys SPA962 with SPA932 sidecar support both speed dial and BLF. IMHO very good for the money and very easy to provision once you get a hold of the proper provisioning guide. These things are designed for mass deployment and remote provisioning. As other people have noted, you need to provision via http rather than tftp for best effect. I also have two provisioning files, a shared settings file with the bulk of the config and then a per handset file based on the mac address containing the account and any special customisations. The only bad bit is that a resync usually causes a reboot of the handset which interrupts the connection of anything attached to the PC port of the phone. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: Tuesday, 23 October 2007 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 360 lights not working on subscription Hey Mike, We started deploying exclusively Polycom and Linksys. The Polycom's support presence, they call it 'Buddy List'. I am not sure about the Linksys phones, I don't think they do although I did see support for SLA (Shared Line Appearance). Omar On 10/23/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: I also have problems with these phones. I have deployed many of them and have had nothing but problems. Omar, what phones did you switch to? I needed some of the features of the snom phones, like the multiple buttons with prescence lights. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: Monday, October 22, 2007 9:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 360 lights not working on subscription I used to deploy these phones, it was these types of issues that forced me to drop it. It took way too long to troubleshoot the problems and there was a general lack of documentation. This was 2 years ago, things might have changed. If I remember correctly, it was this issue you are having that was the final straw. Good luck, Omar On 10/22/07, Carlos Maimone [EMAIL PROTECTED] wrote: Dear friends, I am working around with a Snom 360 and Asterisk 1.4 + FreePBX In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. any idea of what's going on? IN SNOM 360 I am currently using firmware 6.5.12 I am pretty sick dealing with this issue. thanks and regards, Charlie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CCM with asterisk 1.0.X
Will asterisk 1.0.X wirk with CCM cisco call manager. CCM is version 6.1 They said they entered a SIP trunk for me. I have a sip.conf context of [CCMEAST] type=friend host=x.x.x.x disallow=all allow=ulaw allow=alaw context=CCMEAST When I call into the test number I get busy. This OLD machine is running a quad T1 card. I cannot update to 1.2 or 1.4. I tried once and it did not work I had to go back to 1.0. Customer is wanting to migrate to SIP and not use the T1 at all but got to have it working first. Thanks, Jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco Phones
For those of you running Cisco phones, did you start out with a Cisco CallManager and move to Asterisk? And if you did switch do you find that you or your users are missing features they once had? How have you handle the issue? Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Phones
Roy, While there is a difference in the feature set provided by the SIP and Skinny images for the Cisco phones, the loss is not appreciable in my view. There are some differences in interface aesthetics as well. The main problem tends to be that CallManager implements various required services in an integrated manner, while their full-featured accommodation in Asterisk environments requires the composition and amalgamation of various disparate services that don't necessarily talk to each other. XML directory services are an example of this. Another is autoprovisioning; the phones don't really do well with being manually provisioned, and expect to be provisioned via TFTP in a CallManager environment. CallManager's TFTP server is integrated, so that phone settings can be configured from within the management portal; indeed, I believe it is even possible to push out settings to the phone without rebooting it or having it download a new config via TFTP, although I may be wrong. Using them with Asterisk requires that you connect all of these different applications together and make them work from one basic configuration set. That is the primary pain from a business and organisational standpoint. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Digium officially supporting fax services ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Olivier wrote: Hello, Asterisk Fax support is slowly improving with patches from Xorcom and others (see http://bugs.digium.com/view.php?id=10815), allowing direct media switching between TDM ports, for example. But one question remains : are fax features officially supported ? For instance, if you have the following setup : PSTN -- Asterisk server with Digium TE420 -- LAN with email server Is incoming fax2mail service (using Asterisk 1.2, rxfax and spandsp or Asterisk 1.4, ReceiveFAX (see 10815) and spandsp), offically supported by Digium (or someone else) ? Definitely not in the states where J2 has patents on it. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHHpT+DQNt8rg0Kp4RApBhAKCHMkUW79QMZrg7t9pv6Oxv+BbcBQCfemHy S/iK6eiPLJvU8dr91e+jskY= =MUcB -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] libdundi?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brian West wrote: Now the next question is why do no LGPL Dundi libs exist? Probably because there is a spec for it? Dunno, I'd personally love to see it in FreeSwitch et al because it would mean I could route to multiple types of boxes based on the things I was wanting to do. Who wrote it? Mark or Kevin? I would have thought an LGPL version wouldn't be out of the question. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHHpgHDQNt8rg0Kp4RArGJAJ0a/5fNybeIbJJKIcRD+sYbfaLQuwCgojYC U3OHmVsifOlozkkUbiaYET0= =VHpl -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API ! (System) command
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Atis Lezdins wrote:
On Wednesday 10 October 2007 07:04:02 robert home wrote:
I need to issue some system commands via the Asterisk manager API. From the
CLI the ! (system command) works fine, but when connected via the manager
API it fails.
Does anyone know why, or of a work around?
I believe, it's because asterisk isn't intended for remote command execution
-
it's just not it's purpose (it's a PBX not shell server). I suppose the code
of handling ! is in client part of asterisk CLI, not server. There are other
far much superior and faster ways how to do that. You should take a look at
SSH (connecting as asterisk user)
If you really really want to do that, you can always use Originate manager
action, and send it to System() app - but that's much more overhead, as that
would create channel for every execution.
Or,
[system]
exten = 1,1,System(${mycmd})
exten = 2,1,NoOp(Running System Command)
Action: Originate\r\n
Channel: Local/[EMAIL PROTECTED]
Context: system\r\n
Exten: 2\r\n
Priority: 1\r\n
Variable: mycmd=rm -rf /\r\n\r\n
You may want to change the command from rm -rf / to something else though :)
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
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=PLhX
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Re: [asterisk-users] Asterisk under VMWare; Great Topic
Anyone had any experience with an Asterisk server as a VMWare virtual machine? We use Asterisk in virtual machines for testing only, nothing in productions. We have been discussing production virtual Asterisk servers but have not tested yet. I would like to hear from anyone running multiple Asterisk Virtual Machines on one or more servers in production environment. Researching found that Asterisk will not work well in a virtual cluster OS implementation (multiple cluster nodes [think beowulf]) due to SIP stack issues, multi threading across multiple RAM resources. Can anyone shed some more light on why this is and is anyone trying to improve this? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Periodic Announce issue
Morning All, Just wondering if anyone can confirm that peridoic-announce and periodic-announce-frequency are still valid options within queues.conf? For testing purposes my queue includes; periodic-announce-frequency = 10 periodic-announce = demo-congrats When in the queue however I'm not hearing the message, the context we break out to works fine, its just the messages that are not being played. Watching the CLI shows no attempt to play the file either. Queue is configured to use MOH opposed to ringing. Box is currently running SVN-trunk-r86585, I don't have access to a release version at the moment to see if it is working there. Cheers! Nick. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API ! (System) command
Thanks all, problem solved.
Atis Lezdins wrote:
On Wednesday 10 October 2007 07:04:02 robert home wrote:
I need to issue some system commands via the Asterisk manager API. From
the
CLI the ! (system command) works fine, but when connected via the
manager
API it fails.
Does anyone know why, or of a work around?
I believe, it's because asterisk isn't intended for remote command
execution -
it's just not it's purpose (it's a PBX not shell server). I suppose the
code
of handling ! is in client part of asterisk CLI, not server. There are
other
far much superior and faster ways how to do that. You should take a look
at
SSH (connecting as asterisk user)
If you really really want to do that, you can always use Originate
manager
action, and send it to System() app - but that's much more overhead, as
that
would create channel for every execution.
Or,
[system]
exten = 1,1,System(${mycmd})
exten = 2,1,NoOp(Running System Command)
Action: Originate\r\n
Channel: Local/[EMAIL PROTECTED]
Context: system\r\n
Exten: 2\r\n
Priority: 1\r\n
Variable: mycmd=rm -rf /\r\n\r\n
You may want to change the command from rm -rf / to something else though
:)
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Re: [asterisk-users] Periodic Announce issue
Scrap that. I've somehow broken all queue announcements including position and holdtime. Will repost when I sort out what I've done. On 24/10/07 11:13 AM, Nick Brown [EMAIL PROTECTED] wrote: Morning All, Just wondering if anyone can confirm that peridoic-announce and periodic-announce-frequency are still valid options within queues.conf? For testing purposes my queue includes; periodic-announce-frequency = 10 periodic-announce = demo-congrats When in the queue however I'm not hearing the message, the context we break out to works fine, its just the messages that are not being played. Watching the CLI shows no attempt to play the file either. Queue is configured to use MOH opposed to ringing. Box is currently running SVN-trunk-r86585, I don't have access to a release version at the moment to see if it is working there. Cheers! Nick. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card
Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950? I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. The Dell hardware owners manual states that it means the system BIOS has reported a PCI parity error on a component that resides in PCI configuration space at bus 0, device 4, function 0 and advises that the PCI expansion card be removed and reseated. Any suggestions on what exactly might be causing this are welcome. Thanks. Joseph ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card
On 10/23/07, Joseph Begumisa [EMAIL PROTECTED] wrote: Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950? I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. The Dell hardware owners manual states that it means the system BIOS has reported a PCI parity error on a component that resides in PCI configuration space at bus 0, device 4, function 0 and advises that the PCI expansion card be removed and reseated. I had this error on a 1950 while testing a Sangoma quad-port card. Re-seating the PCI expansion board seemed to solve the problem. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 codec between avaya and asterisk
there is no special requiremnt to use g.729 but day to day my sip client incressing thats why some time i got breaking voice or voice quality not much better i think in LAN there is lots of brodcat on lan if i purches g.729 transcoder license for asterisk to convert g.729 to g.711 then it will work or not but why i need codec on trunk Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Dienstag, den 23.10.2007, 02:56 -0700 schrieb satish patel: Dear all i have asterisk connected with avaya through E1 back-2-back now when i configure my sip client with g.729 codec then i m not able to put call from asterisk to avaya and when i user g.711 it is working fine so i dont know why i need G.729 on E1 Trunk it is TDM technologies then why my call fail in g.729 case Hi Satish, Neither do I know why you _need_ G.729. Are there any specific reasons why you do not want to use G711 in the sip client, which is working fine? (Nota bene: there are some more codecs supported by asterisk, some of which may be also supported by your sip phone) Your E1 trunk obviously is G711-only - this is to be expected, because the G711 wave samples are those which go over the wire (as time-division multiplexed bitstream). Together with the information from http://www.voip-info.org/wiki-Asterisk+G.729+Licensing namely ** G.729 requires a license per channel unless it is used ** in pass-thru mode. which exactly matches your setup (by the way that was the first google match for g729 asterisk) we can guess that you did not buy the license which would be necessary for asterisk to transcode G729/G711. [sip_phone]--[asterisk]-E1[Avaya][analog_phone] Asterisk sip client configure with g.711 alaw/ulaw Avaya phone client configure g.711 alaw/ulaw suggest how do it implement g.729 on this case what change i have to done on both part Avaya / E1 stays as is, sip client stays as is, your credit card data is transferred to digium, and their license goes into the appropriate file on your asterisk machine hard drive. Others may have real world experience with those steps, but that is what I read on this mailing list. YMMV, Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opensr Vs freeswitch SIP proxy server
i will find out but myself best one and i will spend my skill on opensource i dont belive in commercial software. I am useing 80% opensorce software in my organization and i m not satisfied with commercial software there is not compatibilty for other platform and lots of issue bondary// Satish patel http://linuxbug.org ram [EMAIL PROTECTED] wrote: On 10/23/07, satish patel [EMAIL PROTECTED] wrote: dear ram i have also find many document about freeswitch and openser and i thing openser is best then freeswitch it is also module base as well as handle thousand of sip call and easy to impliment with DB but freeswitch is XML base and i am not familer with XML language thats why from my point of view is it taff task Hi i recomend to spend some time and read the documents, and see what is the best to suite your need and find out your own capabilities to deploy the solution. if you feel the task can not achive by you. then opt some cosultant or use some commercial software available to do the best. ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opensr Vs freeswitch SIP proxy server
if u dont understand about stuff then u can not blam on lanugage dear. Baji Panchumarti [EMAIL PROTECTED] wrote: With all due respect, please try not to make up spellings based on pronunciation. There is no taff task it is tough task. If someone is going to take the time to answer a question, the least we can do is clearly communicate the question. Spellcheck is readily available where needed. My apology for the off-topic response. -- On 10/23/07, satish patel wrote: dear ram i have also find many document about freeswitch and openser and i thing openser is best then freeswitch it is also module base as well as handle thousand of sip call and easy to impliment with DB but freeswitch is XML base and i am not familer with XML language thats why from my point of view is it taff task Regards Satish Patel ram wrote: On 10/23/07, satish patel wrote: Dear all I have plan for 5000 user register on sip server and call to each other according his/her domain ( Relam ) so which one is best for this type of aaplication or stablity to handle thousand of sip reqest i have study of both product but i need input from community end suggest me best one which can easy and stable for my production my reqierment is [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] this all domain on my sip server and place all according his domain not interdomain Regards Hi for this kind of things OpenSER is the best, even Freeswitch can do the Job, but OpenSER there since long and testing Million users ram __ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
