[Asterisk-Users] sip phone, receiving calls but not placing any call
Hello there, I am configuring a sip-phone, it is receiving calls but its not placing calls. sip debug shows that asterisk received digits from phone. but why its not placing calls please help I have dialed 13 from sip-phone, here is some sip-debug INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.187:5060;branch=z9hG4bKfLZ1GRUt2 Max-Forwards: 70 From: chinee sip:[EMAIL PROTECTED];tag=82veOQ0zKConAx6y To: 13 sip:[EMAIL PROTECTED] Call-ID: y2gsu70CXGySlU0s CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 191 thank you -- Atif ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk, PBX, VoIP and PRI
On Wed, 28 Jul 2004, Chris Johnson wrote: On Wed, 28 Jul 2004, Chris Johnson wrote: Why not plug the PRI into a TE410P in the asterisk box and handle both the ppp and the voice via asterisk? If it'll work, sounds great!! Anyone doing this? Sorry, there are no analog softmodem drivers yet. I have been living in isdn land for too long and forgot about the analog modems. There are indeed drivers for ppp over isdn direct but no softmodem for analog calls. We have solved this by routing those calls to our old pbx (over an E1 PRI) from which a couple of BRI:s go to isdn modems with analog capabilities. So, you should be able to hook up your old PRI equipment to Asterisk, i.e. PSTN -PRI- Asterisk -PRI- Old_equipment \ ---lan--- sip stuff This is sort of what we are doing. As far as I can tell the digital channels are passed transparently from one pri to the other once a call is set up. We can do both analog termination and direct isdn connections. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please share your Solaris experiences on the Asterisk Solaris Wiki page
Okay, will do. So, is there someone I should send patches to or is there a process to get cvs write privileges? - logan On Thu, Jul 29, 2004 at 02:55:08PM +0900, Sunrise Ltd wrote: Logan O'Sullivan Bruns wrote: I know Solaris isn't a well tested platform and I did have to make some minor code changes to get to compile on my sun box. Well done! We need more momentum for Asterisk on non-Linux platforms. Building a community around Solaris much like there is a community around BSD, would be very helpful. This will only happen if Solaris users start sharing their stuff in a place where others can easily find it. So, please share your experiences with the community ... http://www.voip-info.org/tiki-index.php?page=Asterisk+Solaris+Support thanks rgds benjk -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __ GANBARE! NIPPON! Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE http://mail.ganbare-nippon.yahoo.co.jp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Changing Transfer key
Not positive but if you're using SIP, it should be easy to do something
like this in extensions.conf
[office]
exten = *7X.,1,Transfer(${EXTEN:2})
exten = *7X.,2,Congestion()
The Dialplan is not execute WHILE you're on hook.
*THAT* would be a tremendous wishlist of my. Get rid of all the *XX
service codes in chan_zap (e.g. *70, *69, *whatever), put them in a
common file for all channels to use and/or make them usable from the
dialplan.
E.g.:
exten = ~*7XX,1,Transfer(...)
where ~ is the marker that this is an while-the-channel-exists dialplan
entry, listing to in-band/rfc2833/sip-info DTMFs
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RE: [Asterisk-Users] Access voicemail from Cisco 7960
Thanks for your swift reply! It did help me... kind of... ;) Guess what I had to do to get it working on my system? I had to ADD dtmfmode=inband to my config! 8-) But now I have full access to my mailbox! :) Regards, Evert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leif Madsen Sent: 28 July 2004 16:50 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Access voicemail from Cisco 7960 On Wed, 28 Jul 2004 14:22:17 +0200, Evert Meulie [EMAIL PROTECTED] wrote: Hi everyone! Who can tell me how I can access my voicemail? When I dial the voicemail on my Cisco 7960 I get access, but when trying to enter my mailbox number it seems that Asterisk doesn't 'get' any of the keys I press. DTMF problem perhaps? Any suggestions on how/where to fix this...? I had a similar problem. If you look at the console, you'll probably either see it missing digits, or sending too many digits. Even though I was using ulaw as my codec, Asterisk didn't like my specifying dtmfmode=inband. I commented out that line and away it went fine. Here is my current sip.conf for my 7960 which works (connected directly to the Asterisk box) [100] type=friend secret=password username=100 callerid=Leif Madsen 18924 context=extensions ;dtmfmode=inband qualify=yes nat=no host=dynamic canreinvite=no disallow=all allow=ulaw allow=alaw allow=g729 mailbox=100 HTH, Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ToS flags for VoIP
When experimenting with ToS, what would be the most appropriate combination to start with? I'm thinking tos=0x14 should be good in most scenarios, since it combines lowdelay with reliability. Any suggestions? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outgoing works, incoming doesn't...
Addition: the console also has these showing: Jul 29 09:58:06 WARNING[1142106560]: chan_sip.c:612 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Evert Meulie Sent: 28 July 2004 15:28 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Outgoing works, incoming doesn't... Hmm, I get lots of these: to 192.168.2.175:5060 Retransmitting #3 (no NAT): OPTIONS sip:192.168.2.175 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.6:5060;branch=z9hG4bK442bde8b From: asterisk sip:[EMAIL PROTECTED];tag=as6496d70e To: sip:192.168.2.175 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 28 Jul 2004 13:44:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 IP 192.168.2.175 is the phone IP 192.168.11.6 is Asterisk (it's not a routing problem, since other phones on the 192.168.2.x IP's do show up as 'OK') Regards, Evert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: 28 July 2004 15:18 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Outgoing works, incoming doesn't... Evert Meulie wrote: Hi! Problem with my 7960. Outgoing calls work, but incoming don't. A 'sip show peers' gives: Name/usernameHostDyn Nat ACL Mask Port Status 105/105 192.168.2.175D 255.255.255.255 5060 UNREACHABLE Is there something wrong with the config on that phone? If so, who can tell me what? As Asterisk tells you, it's UNREACHABLE from Asterisk. Turn on SIP debug and see what happens - where Asterisk is sending packets and if we get any replies at all. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music On Hold - not working for me... (FIXED)
Hi Eric,
Thanks a bunch for your reply. It certainly pointed me in the right direction!
After a few mins. of investigation I thought it was the -g option, but I was
wrong. Then I was blaming the -c option... also wrong.
Changing my safe_asterisk to have CONSOLE=no instead of CONSOLE=yes actually did
make everything work. Before and after this I tried starting asterisk
manually with -vvv, -vvvd, -vvvgd, -vd, etc. :)
All of these also worked fine! So it really made my wonder... but doing the
following (reformatted) made me realise something else:
# diff -u /usr/src/asterisk/contrib/scripts/safe_asterisk
/usr/sbin/safe_asterisk
--- /usr/src/asterisk/contrib/scripts/safe_asterisk 2004-03-12 21:20:36 +0100
+++ /usr/sbin/safe_asterisk 2004-07-29 10:15:05 +0200
@@ -1,6 +1,6 @@
#!/bin/sh
TTY=9 # TTY (if you want one) for Asterisk to run on
-CONSOLE=yes# Whether or not you want a console
+CONSOLE=no # Whether or not you want a console
[EMAIL PROTECTED] # Who to notify about crashes
DUMPDROP=/tmp
#
@@ -64,18 +64,13 @@
mv /tmp/core ${DUMPDROP}/core.`hostname`-`date -Iseconds`
fi
else
- if [ ${EXITSTATUS} = 0 ]; then
- echo Asterisk ended normally. Aborting.
- exit 0
- else
- echo Asterisk died with code $EXITSTATUS.
- if [ -f /tmp/core ]; then
- mv /tmp/core ${DUMPDROP}/core.`hostname`-`date -Iseconds`
- fi
+ echo Asterisk died with code $EXITSTATUS. Aborting.
+ if [ -f /tmp/core ]; then
+ mv /tmp/core ${DUMPDROP}/core.`hostname`-`date -Iseconds`
fi
+ exit 0
fi
echo Automatically restarting Asterisk.
- sleep 1
done
}
It seems that the script /usr/sbin/safe_asterisk is never replaced when you
make install in asterisk. While it's nice that your own hacks are preserved,
I think it would be a lot nicer to have a warning if that file was updated
since your last install - and it could simply backup your existing
safe_asterisk and replace it with the updated version.
I did not actually find out the real reason why that safe_asterisk script did
not work in the first place. Maybe someone more experienced can answer that
better.
Solution: replacing the safe_asterisk with whatever HEAD version I had
downloaded last made everything (about MOH) work.
Or: Simply run asterisk manually and/or with your own safe wrapper.
Best regards
- avizion
Quoting Hall, Eric M. [EMAIL PROTECTED]:
Have you tried to run * in debug mode? I have the same problem and I
found that if I run * in debug (asterisk -vgcd) mode MOH works. I
have no idea why but that is the only way I can get MOH to work for me.
Good luck and please report back to the list if you find a fix!
-Original Message-
I'm trying to make some simple MOH (Music On Hold) working. So far I've
failed miserably - so I turn here for help.
Basically I've been using the wiki and all the sample confs I could from
there and via google.
SNIP!
--
avizion on irc.freenode.org #asterisk
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[Asterisk-Users] SIP Outbound Proxy Support
In the latest release of chan_sip2 I've added support for SIP Outbound Proxy. I've seen a lot of requests for that lately, so if you can test this and confirm wheather it works for you or not, I'll be grateful. If I get positive reports, we'll try to add this to chan_sip in CVS. It works like this: * Configure outboundproxy in the general section of sip.conf outboundproxy = hostname or IP outboundproxyport = port # (defaults to 5060) All SIP communication are now sent to the proxy IP If you configure localnet= networks, these are excluded, so only outbound traffic goes to the outbound proxy. If this works, we might try to add support for peer-specific outbound proxies to be able to handle FWD and other providers with NAT traversal support through outbound proxies. http://bugs.digium.com/bug_view_page.php?bug_id=759 Enjoy! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compatible E1 card
Hi, all! Are there anyone from Russia? This is quite specific question: are there any _certified_ E1 cards not hard to find, that support EDSS-1 and compatible with Asterisk? I need to gateway E1 coming from PSTN to SIP for large HomeNetwork needs. SIP works well already. Thanks a lot, S. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Changing Transfer key
The Dialplan is not execute WHILE you're on hook. Ahh, my limited english. I meant: The Dialplan is executed while the phone is on-hook. If the phone is off-hook (you're talking), the dialplan doesn't do much. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI theme. If you are interested, please drop me an e-mail. If we get enough speakers, I might to schedule this on the agenda. We already have two parties interested, but I need a few more. I need a reply this week! Today, if possible. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXy config samples
Asterisk refuses to register IAXy.I am using the IAXY Configuration Guide that comes with the IAXy. The guide does not say anything about the [general] section in the iax.conf and the handbook has no IAXy example. Any hints? Thanks Find local movie times and trailers on Yahoo! Movies. http://au.movies.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] is chan_skinny broken?
Matthew Simpson ([EMAIL PROTECTED]) wrote: I am trying to use chan_skinny but when loading the module I get: [ Booting../usr/lib/asterisk/modules/chan_skinny.so: undefined symbol: ast_pickup_call I am using CVS 07/23 I can't get chan_sccp2 to compile, it gives me parse errors, or I'd be using that. :-/ can you try the CVS version of chan_sccp2 (chan-sccp.sf.net). Also what phone are you using ? --jan -- Jan Czmok, Network Engineering Support, Global Access Telecomm, Inc. Ph.: +49 69 299896-35 - fax: +49 69 299896-40 - sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ParkAndAnnounce command !!!
Hi everyone
I'm very, very, very, ... very new to Asterisk and I need some help with the
ParkAndAnnounce command.
Here's what I would like to do. I would like to specify an extension in the
extension.conf file which is using the ParkAndAnnouce command (something
like this)
exten = 200,1,Answer
exten =
200,2,ParkAndAnnounce(PARKED|60|SIP/${EXTEN}|some_context,${EXTEN},1)
exten = 200,3,Hangup
I'm writing an application in JAVA which is using the Manager interface to
manage calls and so on (I have succeeded to hardcode the call on hold and
the transfer and some other actions). In this program I use the Manager
interface to detect calls (I use the channels from the caller side and from
the calling side to put them into a call list). I would like to use the
Manager interface to park the calls from the call list. How can I pass
extensions, or for example, channel parameters to the ParkAndAnnouce command
(is it possible at all?)?
And I would like to know how to use the last part in the ParkAndAnnounce
command (return extension part)
I know the first part is for the messages, the second is the time after the
call is passed to the return extension (the last part), the third part is
about the extension which is used to announce the parked extenision, and the
last part consists of few subparts (context, extension, ) I dont really
understand this last part, the part concerning the return extension.
I had found some example files but I didnt find them very helpful (that's
cause I'm a beginner) :)
If there's another way to park calls, it would be great if you could tell me
about it :)
Thanx for anyone in advance !!!
Greetings from Zolti
_
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[Asterisk-Users] PSTN tone simulation
Hi all is there any way how can I simulate PSTN tone on asterisk. I mean: I take up phone, select number '9' (so I want to call to PSTN) and asterisk change tone to something like . - . - . - Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ToS flags for VoIP
When experimenting with ToS, what would be the most appropriate combination to start with? I'm thinking tos=0x14 should be good in most scenarios, since it combines lowdelay with reliability. Any suggestions? I'm using: tos=0x18 ;sets ip tos bits (=lowdelay, throughput) in sip.conf, however keep in mind the majority of ISP's do not handle any form of qos. If they handle something, its more likely to be either lowdelay or throughput. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] broadvoice/asterisk incoming calls problem attn:Kannaiyan Natesan
Eventhough I don't want to misuse the list, I still want to share information. My *SIP REGISTRATION* goes successfull with asterisk. To be sure, I don't want to touch sip.conf until the udp packets receive my machine. I don't blame asterisk on my part until then. I hope ethereal works in capturing udp packets. I use the command to monitor incoming packets in udp ports. tethereal port 5060 tethereal port 5082 Also the server is in a data center, where it is publically available on internet as I don't need to worry about the problem with NAT. I don't see any incoming packets received on those packets when I receive a call on my broadvoice number. If you still want to touch anything on the sip.conf kindly let me know. -Kannaiyan - Original Message - From: Chris [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 5:34 AM Subject: Re: [Asterisk-Users] broadvoice/asterisk incoming calls problem attn:Kannaiyan Natesan Kannaiyan, If you are not receiving incoming calls, you have * set up incorrectly... Many of us on the list have been using them for some time... it works just fine... If you need help setting it up a lot of people on this list *including myself* would be willing to check out your sip.conf for you and see what you *might* be doing wrong... Also James Jones, from BroadVoice support is a regular on this list and uses * himself... BroadVoice does go down occasionaly but the longest downtime I've experienced from them is 1 day bear in mind that both * and VoIP in general are still relatively new and may not be completely reliable 100% of the time yet... -Chris - Original Message - From: Bartosz Wegrzyn [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 8:14 PM Subject: Re: [Asterisk-Users] broadvoice/asterisk incoming calls problem attn:Kannaiyan Natesan Sir, I do not want to cancel my account with asterisk. I do really trust this company. Especially their support. (JAMES JONES) They do care about asterisk users. Bart, Great. I have got the line a month back. No Incoming calls until now. It was nice to see, Providing you with world class service and value is our mission.. Still waiting to get the support. If you think they are poor and want your money back you can read following, Try BroadVoice service risk-free. If you are dissatisfied, for any reason, you may cancel the service within 30 days of activation and receive a full refund. We'll refund your money hassle-free, with no questions asked. The cancellation procedure is simple - just cancel your service online, send us an e-mail or call and you'll receive an RMA # (Return Merchandise Authorization Number). Please don't get frightened or be in hurry.. Still you have time and broadvoice will take their own time. I recommended to few of my friends, I got the following feedback, I got one broadvoice - 203-***-. Call me there when u find time. It's not good. I'm going to cancel it soon. [Number protected for privacy] You can even analyse with tethereal port 5060 whether you are receiving any packets from broadvoice. I don't receive any of them and I think it is not the fault of asterisk but with broadvoice. -Kannaiyan http://www.goods2world.com - Your VoIP Shop - Original Message - From: Bartosz Wegrzyn [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 12:12 AM Subject: Re: [Asterisk-Users] broadvoice/asterisk incoming calls problem This is what my configuration is: xxx is my phone yyy is my secret [general] externip=lexon.ws port=5060 disallow=all allow=ulaw context=from-broad dtmfmode=inband register = xxx:[EMAIL PROTECTED] tos=0x18 srvlookup=yes [Broadvoice] type=peer username=xxx fromuser=xxx secret=yyy host=sip.broadvoice.com context=from-broad fromdomain=sip.broadvoice.com nat=yes canreinvite=no dtmfmode=inband insecure=yes Incomming calls still fails. NO SOUND AT ALL!!! Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] PSTN tone simulation
Il 12:52, giovedì 29 luglio 2004, [EMAIL PROTECTED] ha scritto: Hi all is there any way how can I simulate PSTN tone on asterisk. I mean: I take up phone, select number '9' (so I want to call to PSTN) and asterisk change tone to something like . - . - . - exten = 9,1,dial(Zap/1/,60) in the context where you are with a zap channel configured (x101p) but I've tryied then I ear the call tone but I can't dial a number. ...so try your own and leave a comment if you succesfully configure it... bye ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy config samples
Asterisk refuses to register IAXy.I am using the IAXY Configuration Guide that comes with the IAXy. The guide does not say anything about the [general] section in the iax.conf and the handbook has no IAXy example. This is my setup: iaxy.conf.sample dhcp codec: ulaw server: 192.168.0.1 user: iaxyuser pass: iaxypass register iax.conf [iaxy] type=friend accountcode=iaxy host=dynamic secret=iaxypass context=demo trunk=no Any hints? Thanks Find local movie times and trailers on Yahoo! Movies. http://au.movies.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
Hi, we are interested. We have developed a comercial web front end for * administrations (mailboxes, voicemail, platform status), as well as a visual tool for dialplan development. I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI theme. If you are interested, please drop me an e-mail. If we get enough speakers, I might to schedule this on the agenda. We already have two parties interested, but I need a few more. I need a reply this week! Today, if possible. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] broadvoice/asterisk incoming calls problem
This is what my configuration is:
xxx is my phone
yyy is my secret
[general]
externip=lexon.ws
port=5060
disallow=all
allow=ulaw
context=from-broad
dtmfmode=inband
register = xxx:[EMAIL PROTECTED]
tos=0x18
srvlookup=yes
[Broadvoice]
type=peer
username=xxx
fromuser=xxx
secret=yyy
host=sip.broadvoice.com
context=from-broad
fromdomain=sip.broadvoice.com
nat=yes
canreinvite=no
dtmfmode=inband
insecure=yes
Incomming calls still fails.
NO SOUND AT ALL!!!
The above [Broadvoice] context with type=peer is generally used
for 'outbound' calls only; something like:
exten = _1.,3,Dial,Sip/Broadvoice/${EXTEN}
However, for inbound calls from Broadvoice, I think you'll need
something like the following in sip.conf:
[sip-broadvoice]
type=user ; handles inbound calls from Broadvoice
context=from-broadvoice
deny=0.0.0.0/0.0.0.0
permit=147.135.8.129/255.255.255.0
permit=147.135.0.129/255.255.255.0
There seems to have been two changes initiated at Broadvoice on
Sunday: 1) Registration, and, 2) no authentication on incoming
calls. (Keep in mind that I just signed up for Broadvoice service
on Saturday, and then experienced the changes/failures on Sunday.)
The majority of discussion and fixes suggested on the list lately
pertains to #1, however a fair number of users have mentioned #2
with very few (if any) responses to those issues.
As I understand #1, the issue is that Broadvoice is providing two
IP addresses with their DNS responses for sip.broadvoice.com, however
asterisk 'always' uses the first entry in the response and never
the second. They might also be using round robin DNS responses,
where in theory their DNS response alternates between two addresses.
Some of the postings have suggested that only one of their two sip
registration servers handle asterisk's registration, and one of the
fixes was to hard code the IP address in /etc/hosts.conf. It sounds
like most folks have worked around the registration issue without
knowing exactly they did (or what additional issues they just added).
The hard coded Ip now limits that * machine to using only one of
the two sip registration servers at Broadvoice, and if that server
happens to be unavailable, * has no where to go.
If anyone has a different interpretation of #1, I'd like to hear it.
Issue #2 is different. Based only on my limited experience from
Saturday (before the changes), incoming * calls from broadvoice
use to include a userid secret to authenticate the session with *.
That appears to have changed on Sunday, and now calls arrive without
the authentication function. Therefore, a section in sip.conf like
the [sip-broadvoice] above that includes type=user is now needed
to handle those calls. If the deny and permit statements are not
included in that context, then calls from any source on the Internet
can be completed via such an open ended context.
There's certainly nothing wrong with allowing such incoming calls
if your dialplan adequately restricts what those calls can reach.
However, if the dialplan allows unrestricted calling, then sooner
or later you might find a hacker making calls through your system.
As I mentioned earlier, I only had a few hours of experience with
the broadvoice config before the changes occurred, so if I've
mis-represented either of the above would someone correct me.
Rich
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Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI theme. No chance for me to pay flight + entry to conference. My wife would hack me in little pieces :-) http://www.holgerschurig.de/destar.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP Soundpoint 600 early dial
Is anyone successfully using this phone with *? I have one, and it is an excellent phone. However, I cannot figure out how to make the phone "early dial" -- that is, automatically dial the number without the user having to press the send button. Any ideas? Thanks, Mike Roberts
Re: [Asterisk-Users] Reverse Battery Disconnect Supervision in X100P or TDM400P FXO
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wed, 28 Jul 2004, Luis Vazquez wrote: Is posible to make the Digium FXO cards detect disconnect supervision by polarity reversal instead of battery drop?? It is possible, but probably not as simple as just detecting the reversal during an active call, since many telcos also signal when the remote end answers by reversing polarity and this might be confused for a hangup. The X100P will not be able to do it from what I understand, but the FXO-modules do detect it. There is no driver support yet. I have some preliminary patches for it (check bug number 9 on http://bugs.digium.com). I have also implemented this for myself in sweden, and it (mostly) works. Find me on IRC i #asterisk (I go by the nick eGnarF) or email me privately and I might be able to help you. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBCOdaWYjaxM2wIe4RAixtAKCZDdZoeXqHO3VsPDV06AwIy7hy2gCfd9e4 fUfK4IRenwO+p6r+p3rwX+A= =CFP+ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
Holger Schurig wrote: What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI theme. No chance for me to pay flight + entry to conference. My wife would hack me in little pieces :-) Me neither... -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP Soundpoint 600 early dial
Mike Roberts wrote: Is anyone successfully using this phone with *? I have one, and it is an excellent phone. However, I cannot figure out how to make the phone early dial -- that is, automatically dial the number without the user having to press the send button. Any ideas? Thanks, Mike Roberts If you access the phone with a web browser, you can add a digitmap in Sip Conf - Local Settings If you have four digit internal numbers, 0 for operator, 9 for outside line: 0[1-8]xxx|9,T etc. The comma just gives you a new dialtone, and the T waits for the timeout you choose as Digitmap Timeout on the same page. This is just an example, you would probably be better off building a more complete digitmap. Regards, Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and RTP / 302 after 18x / Call forwarding after announce
Experts asked now: Is there a way to make this call scenario possible: After an INVITE was received at the asterisk an announcement should be played, then, the caller should be forwarded to another loc. REFER should not be used in any way! I thought about something like this: Client Asterisk --- INVITE > 183 Session Progress RTP Stream [ .. some time .. ] 302 Moved .. Contact: [EMAIL PROTECTED] ACK > But i could not figure out how to make a answer/playback happen without the final (200 ok) response to the INVITE dialog. I thought about patching the chan_sip, but this would take me away from the branch!? Please only answer if: - you know a solution (none sip REFER!) - you may have just an idea (working or not - not important :) ) Sincerely , Michael
Re: [Asterisk-Users] Polycom IP Soundpoint 600 early dial
Hi! You can do this in the web interface sip conf local settings Digitmap You can map the number of digits to be dialed before sending..etc... miklos - Original Message - From: Tor Setane [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 9:26 AM Subject: RE: [Asterisk-Users] Polycom IP Soundpoint 600 early dial Mike Roberts wrote: Is anyone successfully using this phone with *? I have one, and it is an excellent phone. However, I cannot figure out how to make the phone early dial -- that is, automatically dial the number without the user having to press the send button. Any ideas? Thanks, Mike Roberts If you access the phone with a web browser, you can add a digitmap in Sip Conf - Local Settings If you have four digit internal numbers, 0 for operator, 9 for outside line: 0[1-8]xxx|9,T etc. The comma just gives you a new dialtone, and the T waits for the timeout you choose as Digitmap Timeout on the same page. This is just an example, you would probably be better off building a more complete digitmap. Regards, Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] Best Linux for Asterisk
On Wed, Jul 28, 2004 at 10:23:41PM +, Mark Woods said: No, it won't be the absolute latest code, but the Debian community is pretty good about keeping packages updated. ah! ah! ah! really... oh oh, so why debian is eons later in releasing new packages... perhaps you're speaking of -unstable debian... that's wy too unstable. A...but I *am* running unstable! And it's been quite, well, stable! :) There is a huge misconception about stable vs unstable. FWIW, I have found debian unstable to be more stable than most other distro's stable releases. For a truely unstable version, experimental would be it. Most of the unstable behavior has been in GUI based parts: Gnome in particular. Since no sane person runs * on a machine that is also running X, it's a non-issue. I've been running * on unstable for about 6 months now with zero downtime other than a few upgrades. Ditto for about a dozen other servers doing high-volume mail, web serving, etc. I find stable unsuitable for most things as all the packages and libraries are too outdated. Yes, the backports help, but then you are not really running stable anymore are you? There are too many dependancies now on other software that needs to be up to date in order to function properly and have the features needed. Anyway, I don't think that it's possible to have a best linux to run any kind of server on. They are all damn good. The core of any linux distro handling non-gui based server applications is virtually identical. Most of the differences are package versions, minor configuration tweaks, package management, and other non-important (when it comes to stability) factors. Do your own research and find one you are comfortable with. For a platform with long-term stability where packages are not constantly changing, maybe something like WhiteBox Linux which is based on the RedHat Enterprise would be appropriate. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BugetTone Bug Showstopper,
On Wed, 28 Jul 2004 23:31:06 -0400, Seth Remington [EMAIL PROTECTED] wrote: On Wed, 2004-07-28 at 21:00, James Gardiner wrote: How do I get Asterisk to recognise the # key from the granstream phone for doing transfers? Make sure the Grandstream is configured to send DTMF via SIP INFO instead of in-audio. -Seth Also, don't forget to disable the #-key as redial feature. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXy config samples
Thanks Florian, I added line username=iaxyuser in iax.conf and no luck. --- Florian Overkamp [EMAIL PROTECTED] wrote: Hi, -Original Message- Asterisk refuses to register IAXy.I am using the IAXY Configuration Guide that comes with the IAXy. The guide does not say anything about the [general] section in the iax.conf and the handbook has no IAXy example. This is my setup: iaxy.conf.sample dhcp codec: ulaw server: 192.168.0.1 user: iaxyuser pass: iaxypass register iax.conf [iaxy] type=friend accountcode=iaxy host=dynamic secret=iaxypass context=demo trunk=no Any hints? Might help if you set 'username=iaxyuser' in iax.conf ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Find local movie times and trailers on Yahoo! Movies. http://au.movies.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quadbri in NT Mode against PBX.
Hi, We have two QuadBri cards and we want get the following topology: Ericcson - Asterisk (Quad Bri) | Internet We have the interfaces facing the ericcson in NT mode. Before of this change this ericsson was connected withour problem to isdn line from Telefonica of Spain. But with our asterisk we dont have link in the interface in NT mode. We need in this topology the power feed for the quabri? We are using cat5 ethercable for conenct the ericcson with our asterisk. We need an special pinout? The Ericcson works well if we connect their directly to the PSTN. We are located in Spain. I was trying google but we dont have any answer :(. Also I change the signaling and other span parameter without any success. Thks in advance for your help. Do you know this topology or any trick that we need use for resolve this problem with sucess, I check this quadbri connecting the card in TE mode against our isdn lines and works without problem. regards, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Experience with this online seller?
We've only ordered from them once, but so far they have surpassed our experience with other (unnamed) resellers. I placed an order with them for two phones at 4:30pm their time. Within 30 minutes, I had a confirmation invoice and a Fedex tracking number, and the phones went out that night. From other sources, we're about 50%. That means 50% of the time, we get our stuff and the rest of the time the order is either lost or significantly delayed. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 29 Jul 2004, Jean-Yves Avenard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello I'm about to order some few phones from this place: www.thevoipconnection.com Do you guys have any experience with this store? Thank you Regards Jean-Yves -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCH3+XeDVKqIr3GURAs4EAJ4zHpqfAWj5ZmHkg6g/prg5ljAkBQCeIxE1 JqYQcuraeBkWICAFnNwvP4k= =DuVi -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
You dont call this Destar or whatever a GUI Solution for Asterisk, do you? So you dont have to worry about your getting hacked. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Holger Schurig Sent: Thursday, July 29, 2004 7:39 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER * What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI theme. No chance for me to pay flight + entry to conference. My wife would hack me in little pieces :-) http://www.holgerschurig.de/destar.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: [Asterisk-Users] Best Linux for Asterisk
I think same... All distributions are based on same kernels... And in my opinion, Kernel is who does all work in an operative systemm.. I am wrong?... Actually I am running 3 * boxes in 3 Machines with Redhat 9.0, all are Athlon based. I had some problems, but generally those problem was related to bugs on * and not on Linux.. I have some friends that test Asterisk using Gentoo and Debian, with success results... So just select distro what you feel more comfortable... Regards Sebastian Nocetti -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Walt Reed Enviado el: Jueves, 29 de Julio de 2004 10:12 a.m. Para: [EMAIL PROTECTED] Asunto: Re: RE: [Asterisk-Users] Best Linux for Asterisk On Wed, Jul 28, 2004 at 10:23:41PM +, Mark Woods said: No, it won't be the absolute latest code, but the Debian community is pretty good about keeping packages updated. ah! ah! ah! really... oh oh, so why debian is eons later in releasing new packages... perhaps you're speaking of -unstable debian... that's wy too unstable. A...but I *am* running unstable! And it's been quite, well, stable! :) There is a huge misconception about stable vs unstable. FWIW, I have found debian unstable to be more stable than most other distro's stable releases. For a truely unstable version, experimental would be it. Most of the unstable behavior has been in GUI based parts: Gnome in particular. Since no sane person runs * on a machine that is also running X, it's a non-issue. I've been running * on unstable for about 6 months now with zero downtime other than a few upgrades. Ditto for about a dozen other servers doing high-volume mail, web serving, etc. I find stable unsuitable for most things as all the packages and libraries are too outdated. Yes, the backports help, but then you are not really running stable anymore are you? There are too many dependancies now on other software that needs to be up to date in order to function properly and have the features needed. Anyway, I don't think that it's possible to have a best linux to run any kind of server on. They are all damn good. The core of any linux distro handling non-gui based server applications is virtually identical. Most of the differences are package versions, minor configuration tweaks, package management, and other non-important (when it comes to stability) factors. Do your own research and find one you are comfortable with. For a platform with long-term stability where packages are not constantly changing, maybe something like WhiteBox Linux which is based on the RedHat Enterprise would be appropriate. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using round-robin dns for sip registrations
On Thu, 2004-07-29 at 00:36, Greg Hill wrote: I finally decided to get a little source code dirt under my fingernails tonight and dig through chan_sip.c to understand how registrations are currently implemented. The hope is to perhaps at least seed some ideas about how to make registrations to a server name, which resolves to multiple IPs, either attempt each IP in the order they're returned by dns, or, simply attempt to register with them all. This would be a good place for somebody to chime in: which approach would be better? If you haven't already, I would suggest posting this on the developers' list instead. You'll get the response you're looking for there. HTH, Ranbir -- Ranbir Systems Aligned Inc. www.systemsaligned.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queue_log question: which endpoint was connected?
Hello list, as I'm writing a little perl parser for queue_log analysis, I'd like to know *which* telephone answered a specific queue call. Unfortunately app_queue only logs the call id but does not log the call end point. This is okay for SIP endpoints, because their call id is something like SIP/endpointname-1234 so you can reasonably understand who was on answering, but for OH323 I get ID's like OH323/LJ5645 that are meaningless. Is there a way to extract from some other log the fact that OH323/LJ234 was a call placed to - say - OH323/[EMAIL PROTECTED] or can I extract it from some field of the peer data structure queue_log seems to extract data from? (to obtain call id, they gust print peer-name) Any help will be greatly appreciated. Thanks l. -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astricon Conference Call?????????
I know this is probably way out there but Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an interesting proof of concept. I personally am planning on attending, but I know others may not. -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax:(407)682-3455 IAXtel: (700)682-6226 x6543 Cell: (321)229-5311 [EMAIL PROTECTED] www.adstelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using round-robin dns for sip registrations
What about a LVS cluster with persistence? In that way, the phone register always to the same cluster member... And what u are balancing are the # of sip phones a server can handle... On Thu, 29 Jul 2004, Kanwar Ranbir Sandhu wrote: On Thu, 2004-07-29 at 00:36, Greg Hill wrote: I finally decided to get a little source code dirt under my fingernails tonight and dig through chan_sip.c to understand how registrations are currently implemented. The hope is to perhaps at least seed some ideas about how to make registrations to a server name, which resolves to multiple IPs, either attempt each IP in the order they're returned by dns, or, simply attempt to register with them all. This would be a good place for somebody to chime in: which approach would be better? If you haven't already, I would suggest posting this on the developers' list instead. You'll get the response you're looking for there. HTH, Ranbir -- Ranbir Systems Aligned Inc. www.systemsaligned.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
You dont call this Destar or whatever a GUI Solution for Asterisk, do you? Why or whatever? So you dont have to worry about your getting hacked. You can't hack DeStar, because currently it doesn't have any user authentication. If there is no obstacle, nothing can be hacked. But hey, it will get this. Remember, DeStar is still in development. But yes, I aim for DeStar to be an easyly customizable solution. But only when it is finished more then now. Just read TODO.txt and do a grep TODO *.ptl *py file to see what is all open ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incoming caller doesn't hear rining.
Hi, I have an asterisk installation that has been happily working in production for some time (E100P and UK BT ISDN30). Recently I upgraded to HEAD-07/29/04. Now, incoming callers don't hear ringing while calling in. As far as I can tell, my config files haven't changed from what was working before. Can anyone please help before my boss shoots me? JC zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-8 dchan=16 loadzone=uk defaultzone=uk zapata.conf [channels] usecallerid=yes language=en echocancel=yes echocancelwhenbridged=yes rxgain=-5% txgain=+5% immediate=no pridialplan=unknown overlapdial=yes signalling=pri_cpe switchtype=euroisdn context=default group=1 callgroup=1 pickupgroup=1 channel = 1-8 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Conference Call?????????
Steve Woolley wrote: I know this is probably way out there but Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an interesting proof of concept. I personally am planning on attending, but I know others may not. -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax:(407)682-3455 IAXtel: (700)682-6226 x6543 Cell: (321)229-5311 [EMAIL PROTECTED] www.adstelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Awesome idea. -- Joseph Finley Technical Services Manager Professional Receivables Control, Inc. (PRC) S. Arlington Road Akron, Ohio 44312 V: 330.493.9004 X 135 F: 330.493.7123 [EMAIL PROTECTED] This is a private communication. The information contained in this message and its attachments may be privileged and confidential and protected from disclosure. If you are not the intended recipient (or the person responsible for delivery of this message to such person), please do not read, copy and/or deliver or disclose it to others. If you have received this communication in error, please notify us immediately by replying to this message, then destroy all hard copies and delete this communication from your computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Aastra 480e phone ADSI config
There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ; - ; Asterisk default ADSI script ; -; ; Begin with the preamble requirements ; DESCRIPTION Asterisk PBX ; Name of vendor VERSION 0x00 ; Version of stuff ;SECURITY _AST ; Security code SECURITY 0x9BDBF7AC; Security code FDN 0x000F ; Descriptor number In an ADSI script for the 2nd Slot: ; ; Asterisk default ADSI script ; ; ; Begin with the preamble requirements ; DESCRIPTION Asterisk PBX ; Name of vendor VERSION 0x00 ; Version of stuff ;SECURITY _AST ; Security code SECURITY 0x78921D49; Security code FDN 0x85EFD9DA ; Descriptor number ; ; Flags ; FLAG nocallwaiting ; ; Predefined strings ; DISPLAY titles IS -- My PBX -- DISPLAY talkingto IS Call active. JUSTIFY LEFT DISPLAY callname IS $Call1p JUSTIFY LEFT DISPLAY callnum IS $Call1s JUSTIFY LEFT DISPLAY incoming IS Incoming call! JUSTIFY LEFT DISPLAY ringing IS Calling... JUSTIFY LEFT DISPLAY callended IS Call ended. JUSTIFY LEFT DISPLAY missedcall IS Missed call. JUSTIFY LEFT DISPLAY busy IS Busy. JUSTIFY LEFT DISPLAY reorder IS Reorder. JUSTIFY LEFT DISPLAY cwdisabled IS Callwait disabled DISPLAY empty IS asdf ; ; Begin soft key definitions ; KEY callfwd IS CallFwd OR Call Forward OFFHOOK VOICEMODE WAITDIALTONE SENDDTMF *60 GOTO offHook ENDKEY KEY vmail_OH IS VMail OR Voicemail OFFHOOK VOICEMODE WAITDIALTONE SENDDTMF 8500 ENDKEY KEY vmail IS VMail OR Voicemail SENDDTMF 8500 ENDKEY KEY backspace IS BackSpc OR Backspace BACKSPACE ENDKEY KEY cwdisable IS CWDsble OR Disable Call Wait SENDDTMF *70 SETFLAG nocallwaiting SHOWDISPLAY cwdisabled AT 4 TIMERCLEAR TIMERSTART 1 ENDKEY KEY cidblock IS CIDBlk OR Block Callerid SENDDTMF *67 SETFLAG nocallwaiting ENDKEY ; ; Begin main subroutine ; SUB main IS IFEVENT NEARANSWER THEN CLEAR SHOWDISPLAY titles AT 1 NOUPDATE SHOWDISPLAY talkingto AT 2 NOUPDATE SHOWDISPLAY callname AT 3 SHOWDISPLAY callnum AT 4 GOTO stableCall ENDIF IFEVENT OFFHOOK THEN CLEAR CLEARFLAG nocallwaiting CLEARDISPLAY SHOWDISPLAY titles AT 1 SHOWKEYS vmail SHOWKEYS cidblock SHOWKEYS cwdisable UNLESS nocallwaiting GOTO offHook ENDIF IFEVENT IDLE THEN CLEAR SHOWDISPLAY titles AT 1 SHOWKEYS vmail_OH ENDIF IFEVENT CALLERID THEN CLEAR ; SHOWDISPLAY titles AT 1 NOUPDATE ; SHOWDISPLAY incoming AT 2 NOUPDATE SHOWDISPLAY callname AT 3 NOUPDATE SHOWDISPLAY callnum AT 4 ENDIF IFEVENT RING THEN CLEAR SHOWDISPLAY titles AT 1 NOUPDATE SHOWDISPLAY incoming AT 2 ENDIF IFEVENT ENDOFRING THEN SHOWDISPLAY missedcall AT 2 CLEAR SHOWDISPLAY titles AT 1 SHOWKEYS vmail_OH ENDIF IFEVENT TIMER THEN CLEAR SHOWDISPLAY empty AT 4 ENDIF ENDSUB SUB offHook IS IFEVENT FARRING THEN CLEAR SHOWDISPLAY titles AT 1 NOUPDATE SHOWDISPLAY ringing AT 2 NOUPDATE SHOWDISPLAY callname at 3 NOUPDATE SHOWDISPLAY callnum at 4 ENDIF IFEVENT FARANSWER THEN CLEAR SHOWDISPLAY talkingto AT 2 GOTO stableCall ENDIF IFEVENT BUSY THEN CLEAR SHOWDISPLAY titles AT 1 NOUPDATE SHOWDISPLAY busy AT 2 NOUPDATE SHOWDISPLAY callname at 3 NOUPDATE SHOWDISPLAY callnum at 4 ENDIF IFEVENT REORDER THEN CLEAR SHOWDISPLAY titles AT 1 NOUPDATE SHOWDISPLAY reorder AT 2 NOUPDATE SHOWDISPLAY callname at 3 NOUPDATE SHOWDISPLAY callnum at 4 ENDIF ENDSUB SUB stableCall IS IFEVENT REORDER THEN SHOWDISPLAY
[Asterisk-Users] (no subject)
Hi all, I would like to study the asterisk source code(Program). I dont' know from which file i've to start reading the code. can anyone helpme. Regards Shan.
[Asterisk-Users] One More IP Phone for interoperability with Asterisk
Hi, I have been able to use eezeePhone VOIP Phone successfully with asterisk. Here is the config. The config sheet is attached This is phone and the ATA is available soon from http://www.eezeephone.compriced at $75.00 each. Both have SIP+H323 and MGCP (also Net2Phone) compatibility. Seshu Kanuri Title: PA168S V1.36.024 network settingsiptypestaticdhcoemodemppp idppp pinlocal ipsubnet maskrouter ipdnsdns2macprotocol settingsprotocolh323sipmgcpspecialuse serviceservice typecommonmediaringetalkauvtechsubcentrexringtecsmartconddavidacitronasiasoftuptechztehuaweikaimenvoipackasiainfolucentharbouripnyiyangthinkersunteksipphoneinphonexfwdnet2phonestanaphonetxtcservice addr service idnat traversaldisableenablecitronauvtechstunvidaaivgreproxyyiyangtxtcnat addr nat ttlphone numberaccountpinregister portsignal portcontrol port register ttlrtp tosrtp portlocal typephonenumberaccountautomd5 accountcat accountcall typenormalfaststartadvanceddtmfcontrol stringinband audiosignal keypadrfc 2833phone settingsuse dialplandisableenabledialnumprefixhotlinedial numberdddcodeiddcodeiddprefixdddprefixinnerlinedisableenableomit prefixlocal prefixnonlocal prefixanswerring typedtmf0dtmf1dtmf2dtmf3dtmf4dtmf5dtmf6dtmf7dtmf8dtmf9not disturbpcmringuser defineuse digitmapforward number fwd powerofffwd noanswerfwd alwaysfwd busycall waitingaudio settingsaudio typeg729g7231g711ug711aautoaudio framesg.723.1 high ratevadagcaechandset inhandset outspeaker outother settingspasswordsuper passworddebugdisableoutputoutput allremote debugno checksntp ipuse daylightupgrade addr timezone(GMT-12:00)Eniwetok,Kwajalein(GMT-11:00)Midway Island,Samoa(GMT-10:00)Hawaii(GMT-09:00)Alaska(GMT-08:00)Pacific Time(U.S. & Canada)(GMT-07:00)Mountain Time(U.S. & Canada)(GMT-07:00)Arizona(GMT-06:00)Mexico City(GMT-06:00)Saskatchewan(GMT-06:00)Central Time(U.S. & Canada)(GMT-06:00)Central America(GMT-05:00)Bogota,Lima(GMT-05:00)Eastern Time(U.S. & Canada)(GMT-05:00)Indiana(East)(GMT-04:00)Atlantic Time(Canada)(GMT-04:00)Caracas,La Paz(GMT-04:00)Santiago(GMT-03:30)Newfoundland(GMT-03:00)Brasilia(GMT-03:00)Buenos Aires(GMT-03:00)Greenland(GMT-02:00)Mid-Atlantic(GMT-01:00)Cape Verde Is.(GMT-01:00)Azores(GMT)Dublin,Edinburgh,London,Lisbon(GMT)Casablanca,Monrovia(GMT+01:00)Amsterdam,Berne,Rome,Stockholm(GMT+01:00)Belgrade,Budapest(GMT+01:00)Brussels,Copenhagen,Madrid,Paris(GMT+01:00)Sarajevo,Sofija,Warsaw(GMT+01:00)West Central Africa(GMT+02:00)Bucharest(GMT+02:00)Harare(GMT+02:00)Riga(GMT+02:00)Cairo(GMT+02:00)Athens,Istanbul(GMT+02:00)Jerusalem(GMT+03:00)Baghdad(GMT+03:00)Kuwait,Riyadh(GMT+03:00)Moscow,St.Petersburg(GMT+03:00)Nairobi(GMT+03:30)Teheran(GMT+04:00)Abu Dhabi,Muscat(GMT+04:00)Baku(GMT+04:30)Kabul(GMT+05:00)Ekaterinburg(GMT+05:00)Islamabad,Karachi(GMT+05:30)Calcutta,Bombay,New Delhi(GMT+05:45)Katmandu(GMT+06:00)Novosibirsk(GMT+06:00)Dacca(GMT+06:00)Sri Jayawardenepura(GMT+06:30)Rangoon(GMT+07:00)Krasnoyarsk(GMT+07:00)Bangkok,Jakarta,Hanoi(GMT+08:00)Beijing,Hong Kong,Urumqi(GMT+08:00)Kuala Lumpur,Singapore(GMT+08:00)Perth(GMT+08:00)Taipei(GMT+08:00)Ulan Bator(GMT+09:00)Tokyo,Osaka,Sapporo(GMT+09:00)Seoul(GMT+09:00)Yakutsk(GMT+09:30)Adelaide(GMT+09:30)Darwin(GMT+10:00)Brisbane(GMT+10:00)Vladivostok(GMT+10:00)Guam,Port Moresby(GMT+10:00)Hobart(GMT+10:00)Canberra,Melbourne,Sydney(GMT+11:00)Magadan,Sol.Is.(GMT+12:00)Kamchatka,Marshall Is.(GMT+12:00)Wellington,Auckland(GMT+13:00)Nuku'alofaAddress BookUpdate Firmware, Digitmap and Ring
Re: [Asterisk-Users] Best Linux for Asterisk
Sebastian Nocetti [EMAIL PROTECTED] wrote: All distributions are based on same kernels... And in my opinion, Kernel is who does all work in an operative systemm.. I am wrong?... Sort of. libc is the other thing that can affect performance. However, any distribution worth its salt will provide a selection of optimal kernel and libcs because of this. The effort of building a custom kernel and libc is probably worthwhile, but beyond that you should probably spend your efforts elsewhere other than recompiling stuff for the sake of it. Gentoo's performance improvements from recompiling the world are usually more psychological than practical. -- [About a discussion of heavily customised cars.] I thought they were talking about cheap whores - smelly, ugly, brightly coloured, waste of money, and got a cock inside them most of the time. -- Will Hargrave in uknot ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Conference Call?????????
That would be great And for that kind of massive conference attendant, a app for * that can receive an mp3 stream and transfor it as a channel feed for a conference room, would be the solution. I can provide a * server for the astricon with relative high bandwith, for a parallel conference server... On Thu, 29 Jul 2004, Steve Woolley wrote: I know this is probably way out there but Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an interesting proof of concept. I personally am planning on attending, but I know others may not. -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax:(407)682-3455 IAXtel: (700)682-6226 x6543 Cell: (321)229-5311 [EMAIL PROTECTED] www.adstelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Successfully Using $135 Avaya sip phone
OT: how/where can you buy one of these SIP phones? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Conference Call?????????
Steve Woolley wrote: I know this is probably way out there but Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an interesting proof of concept. I personally am planning on attending, but I know others may not. If we can get reliable bandwidth out of the building, I can host such a conference on our network in Chicago. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
Hi Olle- I wonder of you could please post the most recent agenda for each day, even if it's not finalized. Some of us can't attend the whole conference, and so need to pick the best days/times to come. (I'm scheduling a trip, and a stop at astricon could be on the way there) Thanks! Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Thursday, July 29, 2004 2:50 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER * I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI theme. If you are interested, please drop me an e-mail. If we get enough speakers, I might to schedule this on the agenda. We already have two parties interested, but I need a few more. I need a reply this week! Today, if possible. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Conference Call?????????
On Thu, 29 Jul 2004 10:09:59 -0400, Joseph Finley [EMAIL PROTECTED] wrote: Steve Woolley wrote: I know this is probably way out there but Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an interesting proof of concept. I'm pretty sure there are plans for this to happen, but not sure if its a for sure yet (nothing is ever for sure when it comes to planning conferences I'm sure). Maybe Olle will chime in here and let us know if there are still plans for this to happen. Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Winbond drivers
Hi! It is the first time I post to this ML, so please be patient :-) I read that a zaptel driver for Winbond 6692 ISDN adapter is on going ... is there any chance to get some test code ? I just bought a similar adapter: I tried getting a HFC based card, but here in Milan (Italy) ISDN card are getting really uncommon. :-( The home PBX is working fine right now with chan_modem, but I suppose that a zaptel driver would be more efficient. Thank you! Cheers, Giuseppe P.S. Is anybody in the list from Italy ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything seems to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN phone, the call does not get disconnected. My SIP phone goes quiet but doesn't disconnect. If I a few seconds later pick up the PSTN phone again, the connection is still there. Only if I hangup the SIP phone, the call gets destroyed. It seems that Zap doesn't see the remote hangup... Here is my Zaptel config and my Zapata config. I presume the extensions config etc are OK as my call-flow never changed and things were working fine with my AS5300. Am I missing something ? How do I debug the Zap channels ? Cheers, Walter Klomp /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 # This is the line in question... span=2,1,0,ccs,hdb3,crc4 # not used yet span=3,0,0,ccs,hdb3,crc4 # not used yet span=4,0,0,ccs,hdb3,crc4 # not used yet # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-77 dchan=78 bchan=79-93 # Span 4 bchan=94-108 dchan=109 bchan=110-124 alaw=1-124 loadzone=uk defaultzone=uk /etc/asterisk/zapata.conf [channels] context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 signalling=pri_cpe channel = 1-15 channel = 17-31 ; Span 2 group=2 signalling=pri_cpe channel = 32-46 channel = 48-62 ; Span 3 group=3 signalling=pri_cpe channel = 63-77 channel = 79-93 ; Span 4 group=4 signalling=pri_cpe channel = 94-108 channel = 110-124 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] faxing
[EMAIL PROTECTED] wrote: What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate centers. Outgoing has been bad. It seems to work the best if the Sipura user agents have echo cancelation off, but we have to have echo cancelation on our outbound gateways or there is echo in the voice path. Faxing outbound works very rarely, and if it does, it usually can only send a page or two before we get the infamous Poor line condition. Does anyone have a suitable FAX setup working? Using G.711 u or A may work, but don't count on it. Take a look at http://www.opencall.org/faq Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] (no subject)
I guess this is obvious, but you could start with asterisk.c in the asterisk directory. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com http://www.evtmedia.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ShanKutti Sent: Thursday, July 29, 2004 7:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] (no subject) Hi all, I would like to study the asterisk source code(Program). I dont' know from which file i've to start reading the code. can anyone helpme. Regards Shan. http://clients.rediff.com/signature/track_sig.asp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] incoming caller doesn't hear rining.
A very helpful person just sorted the problem out. Apparently, changing the incoming dial in extensions.conf to Tr solved the problem. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 3:09 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] incoming caller doesn't hear rining. Hi, I have an asterisk installation that has been happily working in production for some time (E100P and UK BT ISDN30). Recently I upgraded to HEAD-07/29/04. Now, incoming callers don't hear ringing while calling in. As far as I can tell, my config files haven't changed from what was working before. Can anyone please help before my boss shoots me? JC zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-8 dchan=16 loadzone=uk defaultzone=uk zapata.conf [channels] usecallerid=yes language=en echocancel=yes echocancelwhenbridged=yes rxgain=-5% txgain=+5% immediate=no pridialplan=unknown overlapdial=yes signalling=pri_cpe switchtype=euroisdn context=default group=1 callgroup=1 pickupgroup=1 channel = 1-8 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Experience with this online seller?
I've always had great experience with them. I've had an emergency or two where they were the only ones that had things that I needed in stock and were willing to ship them same day. I'd recommend them 100%. I've also occasionally needed support, and they were manning the online support chat at 8:00 pm one night and were very helpful. Brian On Thu, 29 Jul 2004 06:14:54 -0700 (PDT), Bruce Komito [EMAIL PROTECTED] wrote: We've only ordered from them once, but so far they have surpassed our experience with other (unnamed) resellers. I placed an order with them for two phones at 4:30pm their time. Within 30 minutes, I had a confirmation invoice and a Fedex tracking number, and the phones went out that night. From other sources, we're about 50%. That means 50% of the time, we get our stuff and the rest of the time the order is either lost or significantly delayed. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 29 Jul 2004, Jean-Yves Avenard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello I'm about to order some few phones from this place: www.thevoipconnection.com Do you guys have any experience with this store? Thank you Regards Jean-Yves -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCH3+XeDVKqIr3GURAs4EAJ4zHpqfAWj5ZmHkg6g/prg5ljAkBQCeIxE1 JqYQcuraeBkWICAFnNwvP4k= =DuVi -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One More IP Phone for interoperability with Asterisk
This is phone and the ATA is available soon from http://www.eezeephone.com priced at $75.00 each. This is one more phone based on the PA168 chipset. I guess they're all compatible with Asterisk. I recently added the pages Atron AND PA168 to the wiki. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
Kanuri, Seshu wrote: (B (B You dont call this Destar or whatever (B a "GUI Solution for Asterisk", do you? (B (BDid we or did we not just have a longer discussion about (Bbeing nice to each other on this list after a reminder (Bfrom Mark? (B (BThere are certainly things that can be improved with (BDestar and I am sure Holger will appreciate any (Bconstructive criticism. However, your comment looks more (Blike mud throwing to me, not so nice. (B (Brgds (Bbenjk (B (B (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BGANBARE! NIPPON! (BYahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] One More IP Phone for interoperability with Asterisk
Hi all, This is phone and the ATA is available soon from http://www.eezeephone.com priced at $75.00 each. Both have SIP+H323 and MGCP (also Net2Phone) compatibility. That site's a bit out of whack... Got to the voip phone product page at http://www.eezeephone.com/ezp_frm_productdetails.aspx?product_type=101 and try to match the pictures, descriptions, and prices. Looks like the matching-game homework my kindergartner brings home. If it weren't so dodgy, I might be interested in testing some of those 8x8 sip video phones. ...oooh, and my e-mail to their sales group was just returned as undeliverable -- mailbox full... jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Aastra 480e phone ADSI config
Thanks for your input I managed to get what you suggested going last night and it works fine. I also got a note from Sayson on Commedian Mail. Once I did what they suggested, I got a full Voice mail interface on my phone. Pretty cool! From Sayson If you are using ADSI phones and trying to access Commedian Mail, CM tries to do an FDM download (it's own ADSI script) to the phone first. If you don't change the FDN and secur. code in the CM app, you will get and error. In the app_voicemail.c file (for me it was located in /usr/src/asterisk/apps ), the adsi_begin_download is evoked as follows: if (adsi_begin_download(chan, addesc, adapp, adsec, adver)) Where addapp (fdn) and adsec are hardcoded as follows: static char *adapp = CoMa; static char *adsec = _AST; They need to be changed to the correct FDN and Security numbers for the slot you wish to download. So you don't overwrite your own programming, use slot 3 or four. (I used slot 3 for my sayson 480e) static char *adapp = \xFB\xC6\x45\x0C static char *adsec = \x9B\x60\x94\x30 Then recompile and press the Vmail button on your phone. It should automatically download the script and then you have a bunch of new buttons to play with! On a side note, I am tring to enhance the ADSI programing in the orignal script. Did your supplier give you any help with additional commands etc. I have not found any docs. So far. Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Woolley Sent: Thursday, July 29, 2004 9:14 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Aastra 480e phone ADSI config There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ; - ; Asterisk default ADSI script ; -; ; Begin with the preamble requirements ; DESCRIPTION Asterisk PBX ; Name of vendor VERSION 0x00 ; Version of stuff ;SECURITY _AST ; Security code SECURITY 0x9BDBF7AC; Security code FDN 0x000F ; Descriptor number In an ADSI script for the 2nd Slot: ; ; Asterisk default ADSI script ; ; ; Begin with the preamble requirements ; DESCRIPTION Asterisk PBX ; Name of vendor VERSION 0x00 ; Version of stuff ;SECURITY _AST ; Security code SECURITY 0x78921D49; Security code FDN 0x85EFD9DA ; Descriptor number ; ; Flags ; FLAG nocallwaiting ; ; Predefined strings ; DISPLAY titles IS -- My PBX -- DISPLAY talkingto IS Call active. JUSTIFY LEFT DISPLAY callname IS $Call1p JUSTIFY LEFT DISPLAY callnum IS $Call1s JUSTIFY LEFT DISPLAY incoming IS Incoming call! JUSTIFY LEFT DISPLAY ringing IS Calling... JUSTIFY LEFT DISPLAY callended IS Call ended. JUSTIFY LEFT DISPLAY missedcall IS Missed call. JUSTIFY LEFT DISPLAY busy IS Busy. JUSTIFY LEFT DISPLAY reorder IS Reorder. JUSTIFY LEFT DISPLAY cwdisabled IS Callwait disabled DISPLAY empty IS asdf ; ; Begin soft key definitions ; KEY callfwd IS CallFwd OR Call Forward OFFHOOK VOICEMODE WAITDIALTONE SENDDTMF *60 GOTO offHook ENDKEY KEY vmail_OH IS VMail OR Voicemail OFFHOOK VOICEMODE WAITDIALTONE SENDDTMF 8500 ENDKEY KEY vmail IS VMail OR Voicemail SENDDTMF 8500 ENDKEY KEY backspace IS BackSpc OR Backspace BACKSPACE ENDKEY KEY cwdisable IS CWDsble OR Disable Call Wait SENDDTMF *70 SETFLAG nocallwaiting SHOWDISPLAY cwdisabled AT 4 TIMERCLEAR TIMERSTART 1 ENDKEY KEY cidblock IS CIDBlk OR Block Callerid SENDDTMF *67 SETFLAG nocallwaiting ENDKEY ; ; Begin main subroutine ; SUB main IS IFEVENT NEARANSWER THEN CLEAR SHOWDISPLAY titles AT 1 NOUPDATE SHOWDISPLAY talkingto AT 2 NOUPDATE SHOWDISPLAY callname AT 3 SHOWDISPLAY callnum AT 4 GOTO stableCall ENDIF IFEVENT OFFHOOK THEN CLEAR CLEARFLAG nocallwaiting CLEARDISPLAY SHOWDISPLAY titles AT 1 SHOWKEYS vmail SHOWKEYS cidblock SHOWKEYS cwdisable UNLESS nocallwaiting GOTO offHook ENDIF IFEVENT IDLE THEN CLEAR SHOWDISPLAY titles AT 1 SHOWKEYS vmail_OH ENDIF IFEVENT CALLERID THEN
Re: [Asterisk-Users] Best Linux for Asterisk
Gentoo's performance improvements from recompiling the world are usually more psychological than practical. http://www.funroll-loops.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where to start asterisk sourcecode
Hi all, I would like to study the asterisk source code(Program). I dont' know from which file i've to start reading the code. can anyone helpme. Regards Shan.
RE: [Asterisk-Users] Astricon Conference Call?????????
Hi All! I would be happy to provide our Asterisk server for the users to dial in for the Conference. We have enough bandwidth and the calls should go pretty smooth. Though Offcourse I would need some help in tweaking our server for the load. Let me know if this would be of interest to the team. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen Sent: Thursday, July 29, 2004 10:25 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Astricon Conference Call? On Thu, 29 Jul 2004 10:09:59 -0400, Joseph Finley [EMAIL PROTECTED] wrote: Steve Woolley wrote: I know this is probably way out there but Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an interesting proof of concept. I'm pretty sure there are plans for this to happen, but not sure if its a for sure yet (nothing is ever for sure when it comes to planning conferences I'm sure). Maybe Olle will chime in here and let us know if there are still plans for this to happen. Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *** Asterisk Summer News: The heat is on!
Another issue of Asterisk Summer News, delivered right to your mailbox! Back here in Sweden, it's finally summer weather. Sunshine and some heat. It's good for our ice bears and the snow houses to get some sunshine :-) Asterisk development and IRC chat has gone into a lazy summer mode, but the mailing list is still cooking. It's impossible to keep up with it, for both gurus and newbies, even during summer holidays. This issue will be a short issue with just a few articles. Enjoy! This week's topics: --- * Asterisk 1.0rc1: Feedback, please * Astricon 2004: Early bird discount only applies in July * Asterisk IRC chatters: BEHAVE! * Open Source VoIP Watch: SER 0.8.14 * Dialplan updates: The DIAL() application * Recent CVS changes *** Asterisk 1.0rc1: Feedback, please - So we've had some time to try out the release candidate for Asterisk 1.0. If you haven't tried it yet, please do. It is very important for your business and for the Asterisk community that we try to find and fix as many errors as possible before we release 1.0. With the success and growth we've been experiencing lately in the Asterisk.org project, I believe there will be even more success in the fall. This will certainly lead to more pressure from people that use Asterisk in production. In that situation, we need a stable branch code for production use and a development CVS tree for creative development and dangerous but exciting code. In order to get there, we need your help. Test rc1 (or rc2 which is on it's way) and provide feedback. * Download mirrors: http://www.voip-info.org/wiki-Asterisk-mirrors * Linux RPMs: ftp://ftp.nacs.net/asterisk * Instructions on how to report bugs: http://www.digium.com/bugtracker.html To get better documentation for 1.0, join the asterisk-docs mailing list and contribute to the effort. Leif Madsen and Jared Smith really needs your help in order to get a decent handbook out to the 1.0 release. * http://www.asteriskdocs.org *** Astricon 2004: Early bird discount only applies in July --- Astricon 2004 is getting closer. This is the first Asterisk user's and developer's conference. During July, you will get an early-bird discount on the registration fee so please do not forget to register and pay before july 31. The conference agenda was published this week. Amongst the speakers you'll find: * Mark Spencer, lead developer of Asterisk and founder of Digium * Ravi Sakaria, founder of VoicePulse * David Beckemeyer, Distinguished Research Engineer, Earthlink * Ed Guy, Chief Scientist, Pulver.com Also, a lot of those Asterisk Guru's you find on the IRC channel will speak in live sessions: * bkw_, twisted, blitzrage, jtodd, jsmith You may register for one, two or three days with hotel room booking at the web site. We also have information and discounts on shuttles from the airport. * http://www.astricon.net *** Asterisk IRC chatters: BEHAVE! -- The #asterisk IRC channel have had a tendency to fall into nonsense chatting that has no connection to Asterisk. Also, there's been a number of reports of bad behaviours toward newbie's. This forced Mark Spencer to ask the community to remember that they also have been new to Asterisk and behave friendlier: To everyone who spends time in #asterisk or #asterisk-bugs or basically anything with #asterisk in its name, I want to implore you to please treat new users with respect, and act as good representatives of the Asterisk community. Recently I have had more reports of new users being severely turned off of the project in general due to the comments, reactions and attitudes of a few members of the asterisk channels. The success of the Asterisk project depends upon users and developers, and remember that every one of you, even the most experienced Asterisk users were at one point a newbie and needed some hand holding from someone. Finally, I would also ask that the #asterisk channel in particular please stay as focused on Asterisk related topics as possible. *** Open Source VoIP Watch: SER 0.8.14 -- IPtel.org has released a new stable release of the SIP Express Router, the Open Source SIP Proxy that a lot of commercial service providers use, as well as many companies. They state that 0.8.14 is more of a maintenance release than a release with a lot of new features. So what is the difference between a SIP proxy and Asterisk: * A SIP proxy is never involved in the media stream, it doesn't answer or originate calls * A SIP proxy supports many more SIP applications than voice There are many installations using SER as a SIP Proxy and Asterisk as a feature server for PSTN connectivity, voicemail, conferencing and call center features. Read more * More on Asterisk and SIP Proxy: http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy * Release notes:
[Asterisk-Users] Asterisk and festival
Im having trouble getting festival to work with asterisk. We are running debian (sarge) and got asterisk from CVS. Heres what Im using as far as festival goes Debian (Sarge) gcc version 3.3.4 (Debian 1:3.3.4-3) Connected to Asterisk CVS-HEAD-07/28/04-21:08:19 festival-1.4.3-release.tar.gz speech-tools_1.2.3.orig.tar.gz I got patches for both of these Speech tools patch says __SNIP__ patching file grammar/wfst/wfst_train.cc patching file include/EST_Complex.h patching file include/EST_iostream.h patching file include/EST_THash.h patching file ling_class/EST_relation_aux.cc patching file siod/slib_file.cc patching file speech_class/EST_TrackFile.cc patching file speech_class/EST_wave_cuts.cc patching file speech_class/ssff.cc patching file stats/wagon/dlist.cc patching file stats/wagon/wagon.cc patching file testsuite/hash_regression.cc patching file utils/EST_ServiceTable.cc __SNIP__ and festival patch says __SNIP__ patching file src/modules/base/phrasify.cc patching file src/modules/base/word.cc patching file src/modules/Intonation/int_tree.cc patching file src/modules/Text/token.cc patching file src/modules/Text/xxml.cc patching file src/modules/UniSyn_diphone/us_diphone_index.cc __SNIP__ so I am patching them. I setup and extension to test festival and when I dial it I get __SNIP__ -- Executing Answer(SIP/phone4-17ae, ) in new stack -- Executing Festival(SIP/phone4-17ae, mary had a little lamb) in new stack == Parsing '/etc/asterisk/festival.conf': == Parsing '/etc/asterisk/festival.conf': Found telco-pbx*CLI SIOD ERROR: unbound variable : tts_textasterisk Jul 29 10:59:08 WARNING[1015826]: app_festival.c:440 festival_exec: Festival returned ER == Spawn extension (sip, 555, 2) exited non-zero on 'SIP/phone4-17ae' __SNIP__ Any ideas? Thanks in advance Adam
Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
Scott Stingel wrote: Hi Olle- I wonder of you could please post the most recent agenda for each day, even if it's not finalized. Some of us can't attend the whole conference, and so need to pick the best days/times to come. (I'm scheduling a trip, and a stop at astricon could be on the way there) The most recent agenda is always on the web site - you have all the details there and I update as soon as I know there's a change. http://www.astricon.net /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] faxing
We have at least 3 customers with Cisco ATA186's plugged into a fax machine. They can send and receive faxes perfectly. The config in Asterisk is no different than any other ATA186. G711Ulaw is the codec we use. Supposedly the Sipura SP-2000 we're now using can do faxes as well. Haven't tested it yet. - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 10:44 AM Subject: Re: [Asterisk-Users] faxing [EMAIL PROTECTED] wrote: What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate centers. Outgoing has been bad. It seems to work the best if the Sipura user agents have echo cancelation off, but we have to have echo cancelation on our outbound gateways or there is echo in the voice path. Faxing outbound works very rarely, and if it does, it usually can only send a page or two before we get the infamous Poor line condition. Does anyone have a suitable FAX setup working? Using G.711 u or A may work, but don't count on it. Take a look at http://www.opencall.org/faq Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] broadvoice/asterisk incoming calls problem
Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 29, 2004 5:02 AM
Subject: Re: [Asterisk-Users] broadvoice/asterisk incoming calls problem
This is what my configuration is:
xxx is my phone
yyy is my secret
[general]
externip=lexon.ws
port=5060
disallow=all
allow=ulaw
context=from-broad
dtmfmode=inband
register = xxx:[EMAIL PROTECTED]
tos=0x18
srvlookup=yes
[Broadvoice]
type=peer
username=xxx
fromuser=xxx
secret=yyy
host=sip.broadvoice.com
context=from-broad
fromdomain=sip.broadvoice.com
nat=yes
canreinvite=no
dtmfmode=inband
insecure=yes
Incomming calls still fails.
NO SOUND AT ALL!!!
The above [Broadvoice] context with type=peer is generally used
for 'outbound' calls only; something like:
exten = _1.,3,Dial,Sip/Broadvoice/${EXTEN}
However, for inbound calls from Broadvoice, I think you'll need
something like the following in sip.conf:
[sip-broadvoice]
type=user ; handles inbound calls from Broadvoice
context=from-broadvoice
deny=0.0.0.0/0.0.0.0
permit=147.135.8.129/255.255.255.0
permit=147.135.0.129/255.255.255.0
There seems to have been two changes initiated at Broadvoice on
Sunday: 1) Registration, and, 2) no authentication on incoming
calls. (Keep in mind that I just signed up for Broadvoice service
on Saturday, and then experienced the changes/failures on Sunday.)
The majority of discussion and fixes suggested on the list lately
pertains to #1, however a fair number of users have mentioned #2
with very few (if any) responses to those issues.
As I understand #1, the issue is that Broadvoice is providing two
IP addresses with their DNS responses for sip.broadvoice.com, however
asterisk 'always' uses the first entry in the response and never
the second. They might also be using round robin DNS responses,
where in theory their DNS response alternates between two addresses.
Some of the postings have suggested that only one of their two sip
registration servers handle asterisk's registration, and one of the
fixes was to hard code the IP address in /etc/hosts.conf. It sounds
like most folks have worked around the registration issue without
knowing exactly they did (or what additional issues they just added).
The hard coded Ip now limits that * machine to using only one of
the two sip registration servers at Broadvoice, and if that server
happens to be unavailable, * has no where to go.
If anyone has a different interpretation of #1, I'd like to hear it.
Issue #2 is different. Based only on my limited experience from
Saturday (before the changes), incoming * calls from broadvoice
use to include a userid secret to authenticate the session with *.
That appears to have changed on Sunday, and now calls arrive without
the authentication function. Therefore, a section in sip.conf like
the [sip-broadvoice] above that includes type=user is now needed
to handle those calls. If the deny and permit statements are not
included in that context, then calls from any source on the Internet
can be completed via such an open ended context.
There's certainly nothing wrong with allowing such incoming calls
if your dialplan adequately restricts what those calls can reach.
However, if the dialplan allows unrestricted calling, then sooner
or later you might find a hacker making calls through your system.
As I mentioned earlier, I only had a few hours of experience with
the broadvoice config before the changes occurred, so if I've
mis-represented either of the above would someone correct me.
Rich
H as for #2, I've NEVER used authentication on my inbound calls, even
before they made the change and it has always worked... I use a bogus
context as my default context (I don't want unrouted calls) and I've set up
the correct context for my broadvoice-incoming contexts.. As far as
INSECURE=YES or VERY, I don't use that at all and it still works fine...
I'm a little concerned by the NAT=YES in the sip.conf sample below. That
could be the culprit for the no audio problem right there, especially if the
RTP stream doesn't know where to go... If you really really MUST use NAT,
why not try using port forwarding on your router and forward UDP port 5060
and the RTP ports that you have set in your rtp.conf directly to your *
server, then set NAT=NO or NAT=NEVER and try that...
Also I don't see any localnet entries or externip entries... if you're using
nat, you kinda need those...
NAT is evil and should be destroyed and sent back to hell from whence it
came...
Here's a sample of my sip.conf in case this helps you...
[general]
port=5060
bind=0.0.0.0
externip=24.20.x.x (why not try using an ip instead of a hostname...)
localnet=10.100.5.0/24 (not sure if this is needed anymore...)
context=bogus
srvlookup=yes
tos=0x18
maxexpirey=3600
defaultexpirey=120
progressinband=no
disallow=all
allow=gsm
allow=alaw
allow=ulaw
allow=adpcm
allow=speex
RE: [Asterisk-Users] IAXy config samples
Hi, -Original Message- I added line username=iaxyuser in iax.conf and no luck. Odd. Is the line in the same section as the rest of the config ? With that kind of setup I have it working properly. You might want to try 'iax2 debug' and possibly tcpdump to see if any traffic is exchanged. Other than that I'm out of options. Sounds like a misconfiguration or typo to me. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] incoming caller doesn't hear rining.
Johan wrote: A very helpful person just sorted the problem out. Apparently, changing the incoming dial in extensions.conf to Tr solved the problem. T doesn't do anything for ringing only r does. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
This is quite common in some countries. Analogue lines are some times configured for 'calling party clearing', where an inbound call to an analogue line will hold the line for some minutes before timing out. Might this explain the behaviour? Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walter Klomp Sent: 29 July 2004 15:44 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything seems to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN phone, the call does not get disconnected. My SIP phone goes quiet but doesn't disconnect. If I a few seconds later pick up the PSTN phone again, the connection is still there. Only if I hangup the SIP phone, the call gets destroyed. It seems that Zap doesn't see the remote hangup... Here is my Zaptel config and my Zapata config. I presume the extensions config etc are OK as my call-flow never changed and things were working fine with my AS5300. Am I missing something ? How do I debug the Zap channels ? Cheers, Walter Klomp /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 # This is the line in question... span=2,1,0,ccs,hdb3,crc4 # not used yet span=3,0,0,ccs,hdb3,crc4 # not used yet span=4,0,0,ccs,hdb3,crc4 # not used yet # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-77 dchan=78 bchan=79-93 # Span 4 bchan=94-108 dchan=109 bchan=110-124 alaw=1-124 loadzone=uk defaultzone=uk /etc/asterisk/zapata.conf [channels] context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 signalling=pri_cpe channel = 1-15 channel = 17-31 ; Span 2 group=2 signalling=pri_cpe channel = 32-46 channel = 48-62 ; Span 3 group=3 signalling=pri_cpe channel = 63-77 channel = 79-93 ; Span 4 group=4 signalling=pri_cpe channel = 94-108 channel = 110-124 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unauthenticated calls from a specific IP
We put a VWIC and a DSP in a Cisco 1720. The purpose will be for a customer to use a T1 Crossover cable to connect the 1720 into their existing PBX system. It'll be a "Virtual T1 PRI" type of thing. The Cisco 1720 will make the conversion to SIP and send it to our Asterisk server. As far as his PBX is concerned, it's talking to a standard T1 PRI from the local telco or whatever. The issue is Cisco routers don't support SIP registration/authentication. I want this customer to be in his own context in the extensions.conf file. What I was thinking is, if I remove "username" and "secret" from the sip.conf for astandard userentry, but do a "context=whatever and a "host=x.x.x.x" for his specific IP, if an unauthenticated request comes in from that IP it should automatically put him in that context, instead of the default one specific at the top of the file in the [general] section.Also, if I forward several DID's to SIP/customer1 (customer1 being what I put in brackets for this entry, ie [customer1]) it should see the host=x.x.x.x and send it to that IP, regardless of authentication. sip.conf Example below: [customer1]context=customer1contexttype=friendqualify=nohost=x.x.x.xcanreinvite=nodtmfmode=inbandnat=nocallerid="Customer 1" 1235551212accountcode=8785amaflags=billinginsecure=very extensions.conf Example below: [incoming] exten = 1235551212,1,Goto(customer1context,1235551212,1) exten = 1235551213,1,Goto(customer1context,1235551213,1) exten = 1235551214,1,Goto(customer1context,1235551214,1) [customer1context] include = outgoing_local include = outgoing_longdistance include = outgoing_international exten = 1235551212,1,Dial(SIP/customer1,30,r) exten = 1235551213,1,Dial(SIP/customer1,30,r) exten = 1235551214,1,Dial(SIP/customer1,30,r) Maybe I should put a "defaultip=x.x.x.x" in the sip.conf section as well? Will this work? Thanks, Deon 550 Fairway DriveSuite 210Deerfield Beach, FL 33441Online: www.webunited.net Deon Rodden Toll Free: 1-877-538-5969 x 208Phone: 954-418-8884 x 208Fax: 954-418-8635E-mail: [EMAIL PROTECTED]
Re: [Asterisk-Users] One More IP Phone for interoperability with Asterisk
Holger Schurig wrote: This is phone and the ATA is available soon from http://www.eezeephone.com priced at $75.00 each. This is one more phone based on the PA168 chipset. I guess they're all compatible with Asterisk. I recently added the pages Atron AND PA168 to the wiki. How did you work that out? All the IP phones I can see there are video phones. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quadbri in NT Mode against PBX.
Hi, if you connect the Ericcson to the Asterisk you need a crossover-cable. RJ-45 TE NT Polarity 3 Transmit Receive+ 4 Receive Transmit + 5 Receive Transmit- 6 Transmit Receive - Marco - Original Message - From: Daniel Concepcion [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 3:14 PM Subject: [Asterisk-Users] Quadbri in NT Mode against PBX. Hi, We have two QuadBri cards and we want get the following topology: Ericcson - Asterisk (Quad Bri) | Internet We have the interfaces facing the ericcson in NT mode. Before of this change this ericsson was connected withour problem to isdn line from Telefonica of Spain. But with our asterisk we dont have link in the interface in NT mode. We need in this topology the power feed for the quabri? We are using cat5 ethercable for conenct the ericcson with our asterisk. We need an special pinout? The Ericcson works well if we connect their directly to the PSTN. We are located in Spain. I was trying google but we dont have any answer :(. Also I change the signaling and other span parameter without any success. Thks in advance for your help. Do you know this topology or any trick that we need use for resolve this problem with sucess, I check this quadbri connecting the card in TE mode against our isdn lines and works without problem. regards, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
Benj, (B (BThis is not mud throwing. This is constructive criticism for the contributors to be (Bmore productive. (B (BPlease remember that anything that is given here as "usable solution" should not only (Bbe usable but also useful. Whereas Destar is not. It just does not do anything. (BIt has a totally blank vanilla interface to it. (B (BRemember that the issue here is "GUI Solution for Managing Asterisk". Destar does not (Bfall into that category and the code written in Python just dont jell well with the (BAsterisk community, who are mostly Non-Techie novices trying to understand what is (Bgood about Asterisk. (B (BI made these comments as I have spent some time trying to figure out if Destar is (Breally useful. (B (BSeshu Kanuri (B (B-Original Message- (BFrom: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Sunrise Ltd (BSent: Thursday, July 29, 2004 10:56 AM (BTo: astusr (BSubject: Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER * (B (B (BKanuri, Seshu wrote: (B (B You dont call this Destar or whatever (B a "GUI Solution for Asterisk", do you? (B (BDid we or did we not just have a longer discussion about (Bbeing nice to each other on this list after a reminder (Bfrom Mark? (B (BThere are certainly things that can be improved with (BDestar and I am sure Holger will appreciate any (Bconstructive criticism. However, your comment looks more (Blike mud throwing to me, not so nice. (B (Brgds (Bbenjk (B (B (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BGANBARE! NIPPON! (BYahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] incoming caller doesn't hear rining.
On 29/07/2004 at 15:49 Johan wrote: A very helpful person just sorted the problem out. Apparently, changing the incoming dial in extensions.conf to Tr solved the problem. Thanks ...but your caller will get a ringing tone even if your phone number is engaged... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to start asterisk sourcecode
On 29 Jul 2004, ShanKutti wrote: I would like to study the asterisk source code(Program). I dont' know from which file i've to start reading the code. can anyone helpme. depends on what you're trying to do, I guess.. if you want to start at the entry point of the asterisk binary, then 'grep main( *' indicates that asterisk.c might be a good place to begin. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux for Asterisk
Exactly...which is why I run unstable. -Mark perhaps you're speaking of -unstable debian... that's wy too unstable. A...but I *am* running unstable! And it's been quite, well, stable! :) There is a huge misconception about stable vs unstable. FWIW, I have found debian unstable to be more stable than most other distro's stable releases. For a truely unstable version, experimental would be it. Most of the unstable behavior has been in GUI based parts: Gnome in particular. Since no sane person runs * on a machine that is also running X, it's a non-issue. I've been running * on unstable for about 6 months now with zero downtime other than a few upgrades. Ditto for about a dozen other servers doing high-volume mail, web serving, etc. I find stable unsuitable for most things as all the packages and libraries are too outdated. Yes, the backports help, but then you are not really running stable anymore are you? There are too many dependancies now on other software that needs to be up to date in order to function properly and have the features needed. Anyway, I don't think that it's possible to have a best linux to run any kind of server on. They are all damn good. The core of any linux distro handling non-gui based server applications is virtually identical. Most of the differences are package versions, minor configuration tweaks, package management, and other non-important (when it comes to stability) factors. Do your own research and find one you are comfortable with. For a platform with long-term stability where packages are not constantly changing, maybe something like WhiteBox Linux which is based on the RedHat Enterprise would be appropriate. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] One More IP Phone for interoperability with Asterisk
This is a Beta site in development. The pictures and products are not real. The site is Not up yet. Hold on to your mudbricks for a few days. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy Jones Sent: Thursday, July 29, 2004 11:00 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] One More IP Phone for interoperability with Asterisk Hi all, This is phone and the ATA is available soon from http://www.eezeephone.com priced at $75.00 each. Both have SIP+H323 and MGCP (also Net2Phone) compatibility. That site's a bit out of whack... Got to the voip phone product page at http://www.eezeephone.com/ezp_frm_productdetails.aspx?product_type=101 and try to match the pictures, descriptions, and prices. Looks like the matching-game homework my kindergartner brings home. If it weren't so dodgy, I might be interested in testing some of those 8x8 sip video phones. ...oooh, and my e-mail to their sales group was just returned as undeliverable -- mailbox full... jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Motorola MTA RFC3389 Problems
Testing MTA VT1005, one specific issue, RFC3389 support incomplete, please turn off on client... I 've tried all sorts of RTP settings (there's only 4 possible options). And no luck... Any help? Luke ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
On Thu, 29 Jul 2004, Walter Klomp wrote: However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN phone, the call does not get disconnected. My SIP phone goes quiet but doesn't disconnect. If I a few seconds later pick up the PSTN phone again, the connection is still there. Only if I hangup the SIP phone, the call gets destroyed. It seems that Zap doesn't see the remote hangup... Normally a hangup at the b-subscriber (the receiving end) does not tear down the call immediatly, at least not for analog lines from the incumbent operator here in Sweden. I think it is something like 10-30s until the call is released in that case. Did you call an analog phone and how long did you leave it on hook? Am I missing something ? How do I debug the Zap channels ? You need to set up debugging in the corresponding conf file for Asterisk. Debugging of the PRI signalling is then set up with pri debug span ??? or pri intense debug span ??? Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: [EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One More IP Phone for interoperability with Asterisk
Is that Video phone really only $200? And it's SIP compatible with any Asterisk server? Packet 8's was interesting but I never wanted packet 8 service, want to use my own server. Looks like the phone is only $200? - Original Message - From: Jeremy Jones [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 10:59 AM Subject: RE: [Asterisk-Users] One More IP Phone for interoperability with Asterisk Hi all, This is phone and the ATA is available soon from http://www.eezeephone.com priced at $75.00 each. Both have SIP+H323 and MGCP (also Net2Phone) compatibility. That site's a bit out of whack... Got to the voip phone product page at http://www.eezeephone.com/ezp_frm_productdetails.aspx?product_type=101 and try to match the pictures, descriptions, and prices. Looks like the matching-game homework my kindergartner brings home. If it weren't so dodgy, I might be interested in testing some of those 8x8 sip video phones. ...oooh, and my e-mail to their sales group was just returned as undeliverable -- mailbox full... jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
Hi, in Spain that process is correct. If you setup a communication between a caller and a called, if called phone hangs, in caller side hear a silence, but is a correct process. It's is due to in the called side you can hangup a phone and pickup other phone without lost communication. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Walter Klomp Enviado el: jueves, 29 de julio de 2004 16:44 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything seems to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN phone, the call does not get disconnected. My SIP phone goes quiet but doesn't disconnect. If I a few seconds later pick up the PSTN phone again, the connection is still there. Only if I hangup the SIP phone, the call gets destroyed. It seems that Zap doesn't see the remote hangup... Here is my Zaptel config and my Zapata config. I presume the extensions config etc are OK as my call-flow never changed and things were working fine with my AS5300. Am I missing something ? How do I debug the Zap channels ? Cheers, Walter Klomp /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 # This is the line in question... span=2,1,0,ccs,hdb3,crc4 # not used yet span=3,0,0,ccs,hdb3,crc4 # not used yet span=4,0,0,ccs,hdb3,crc4 # not used yet # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-77 dchan=78 bchan=79-93 # Span 4 bchan=94-108 dchan=109 bchan=110-124 alaw=1-124 loadzone=uk defaultzone=uk /etc/asterisk/zapata.conf [channels] context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 signalling=pri_cpe channel = 1-15 channel = 17-31 ; Span 2 group=2 signalling=pri_cpe channel = 32-46 channel = 48-62 ; Span 3 group=3 signalling=pri_cpe channel = 63-77 channel = 79-93 ; Span 4 group=4 signalling=pri_cpe channel = 94-108 channel = 110-124 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] faxing
What's wrong with ftp://ftp.opencall.org/pub/ It says Can't open data connection On Thu, 2004-07-29 at 17:44, Steve Underwood wrote: [EMAIL PROTECTED] wrote: What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate centers. Outgoing has been bad. It seems to work the best if the Sipura user agents have echo cancelation off, but we have to have echo cancelation on our outbound gateways or there is echo in the voice path. Faxing outbound works very rarely, and if it does, it usually can only send a page or two before we get the infamous Poor line condition. Does anyone have a suitable FAX setup working? Using G.711 u or A may work, but don't count on it. Take a look at http://www.opencall.org/faq Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards Vlad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Conference Call?????????
Is it only the guest speakers your interested in listening to? Or is it specific vendos as they show off their products or enhancements to the crowd? With so much noise and people talking at Astricon, how do people in the conference expect to hear any one conversation, or one topic? However, I would be willing to dedicate some bandwidth and an Asterisk server to assist with this cause, I can assign it a South Florida DID as well. We're an ISP w/ several DS3's and a 100mb NMLI. - Original Message - From: Kanuri, Seshu [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 11:21 AM Subject: RE: [Asterisk-Users] Astricon Conference Call? Hi All! I would be happy to provide our Asterisk server for the users to dial in for the Conference. We have enough bandwidth and the calls should go pretty smooth. Though Offcourse I would need some help in tweaking our server for the load. Let me know if this would be of interest to the team. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen Sent: Thursday, July 29, 2004 10:25 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Astricon Conference Call? On Thu, 29 Jul 2004 10:09:59 -0400, Joseph Finley [EMAIL PROTECTED] wrote: Steve Woolley wrote: I know this is probably way out there but Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an interesting proof of concept. I'm pretty sure there are plans for this to happen, but not sure if its a for sure yet (nothing is ever for sure when it comes to planning conferences I'm sure). Maybe Olle will chime in here and let us know if there are still plans for this to happen. Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astricon Recordings?
Just a died question. Will all of the sessions be recorded and made available? Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Thursday, July 29, 2004 10:34 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER * Scott Stingel wrote: Hi Olle- I wonder of you could please post the most recent agenda for each day, even if it's not finalized. Some of us can't attend the whole conference, and so need to pick the best days/times to come. (I'm scheduling a trip, and a stop at astricon could be on the way there) The most recent agenda is always on the web site - you have all the details there and I update as soon as I know there's a change. http://www.astricon.net /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
Walter Klomp wrote: Hi [snip] However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN phone, the call does not get disconnected. My SIP phone goes quiet but doesn't disconnect. If I a few seconds later pick up the PSTN phone again, the connection is still there. Only if I hangup the SIP phone, the call gets destroyed. It seems that Zap doesn't see the remote hangup... [snip] If memory serves me well (moved back to DK a year ago) then this is normal Singtel behaviour for subscriber-to-subscriber calling (it's so you can hang-up and go to another room, pick-up and continue). How long time before you see a hangup if you leave the PSTN side on-hook after the call ?? -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice problems again Attn: James
I can attest that my SUPPORTED device never lost service. This was a Cisco 7960 configured with their TFTP config file (although from my tftp server, I intercepted the file before allowing the phone to use it, but didn't make any changes). It uses the sip server sip.broadvoice.com via the SRV records for the outbound sip proxy listed as as proxy.broadvoice.com So, I chalked it up to confusion, they changed the rules and prevented direct connections to their sip server. Their supported devices worked fine using their configuration because they the outbound proxy. I will also admit that using a single phone on their system is totally useless, but I plugged it in to try and narrow down the problem, within hours of the outage start on Sunday evening. Charlie Hedlin Brian McManus wrote: I do have a few questions about the broadvoice outage: 1) Did supported devices ever loose service? 2) IDo supported devices use the same SIP server, aka, the SIP SRV entries in sip.broadvoice.com, as asterisk would have if configured properly? 3) Would the asterisk SIP SRVLOOKUP entry prevent non-supported devices from loosing service during that outage? I'm only curious because I may be giving them service, as they are the only VoIP provider with number portability that will allow asterisk, and third party SIP devices. However, long outages are of course unacceptable If there supported devices never lost service, i'm willing to chalk it up to a little confusion, and assume that properly configuring DNS SRV Lookups is how I will prevent such an outage in the future. Brian James Jones wrote: not sure I know is pinging does not work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wolfgang S. Rupprecht Sent: Monday, July 26, 2004 5:00 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Broadvoice problems again Attn: James [EMAIL PROTECTED] (James Jones) writes: you can not ping that address because ICMP is turned off. Do you mean *all* ICMP is turned off or just icmp-echo-request / icmp-echo-reply? -wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice mail problem
I am having a problem with getting voice mails, even when the caller hangs up before getting to the recording prompt. If I call my number, even if I hang up the second I get the I'm not in recording, it still generates a voicemail. Is there a way around this? Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] incoming caller doesn't hear rining.
Yes, that is correct. For some reason, for the last few months, I haven't needed the r for incoming calls to hear ringing. Somthing apparently changed at some point. Johan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Thursday, July 29, 2004 4:42 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] incoming caller doesn't hear rining. Johan wrote: A very helpful person just sorted the problem out. Apparently, changing the incoming dial in extensions.conf to Tr solved the problem. T doesn't do anything for ringing only r does. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Conference Call?
All you need is enough bandwidth to upstream one good signal, the users on this list willing to donate bandwidth and equipment can then redistribute it to the others. I don't think Dial up is a very good idea, but having access to a shared T1 or even wireless internet access may be a possibility. Another thing you could do is use a regular phone to call into a DID and enter the conference, then everybody can join that conference and listen. No bandwidth required, just a phone call to the distributor's Asterisk server. Then just keep that phone near the person speaking, like a microphone. - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 11:38 AM Subject: Re: [Asterisk-Users] Astricon Conference Call? If possible, we will broadcast the Asterisk Developer's meeting on the friday. Internet connections in conference hotels is a complex and utterly commercial story, where it is easy to reach sales, but not easy to reach someone that has a clue. We don't know what we can do, what the specifics are in regards to NAT and FW is and the available bandwidth... As soon as we know, we'll inform the community. This also applies to WLAN in conference rooms :-) /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Experience with this online seller?
On Thu, 29 Jul 2004 14:33:02 +1000, Jean-Yves Avenard [EMAIL PROTECTED] wrote: Hello I'm about to order some few phones from this place: www.thevoipconnection.com Do you guys have any experience with this store? Thank you Regards Jean-Yves I've ordered a couple Budgetone 101's from them; their service is top notch, though I've never had to try their tech-support. Shipping was fast and reasonably priced, and they seem to have the best price on Budgetone's on the net. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Aastra 480e phone ADSI config
On Thu, 2004-07-29 at 11:04, Martin Keding wrote: On a side note, I am tring to enhance the ADSI programing in the orignal script. Did your supplier give you any help with additional commands etc. I have not found any docs. So far. There aren't any other commands that Asterisk's implementation of ADSI supports other than the ones that are demonstrated in the asterisk.adsi script. That's not to say that ADSI doesn't have more commands, just that Asterisk does not yet support them. If you take a peek at the code in app_adsiprog.c you will see all the commands that are supported sprinkled through the code (specifically the adsi_process() function). -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] One More IP Phone for interoperability with Asterisk
Those are test pages. Not real. The site is still in development and testing. This IP Phone is not listed yet. But here is the description of the same which is still in beta. I got this phone from other sources(dont ask me how). Hi Folks! Netweb Group, Inc. is happy to announce the immediate availability of eezeephone *Ship that has great features for Asterisk Connectivity. *Ship Operates on H323 as well as SIP and MGCP transparently. *Ship can be set with an E.164 number so that it operates like a regular phone on the network using Asterisk connectivity. Two 10 base-TRJ-45 port for connecting with network are available, one for the PC and another for the Phone. Good interoperability with IP phone, gateway and gatekeeper, which are compatible with H323V4. It can easily integrate with any other SoftPhone or PSTN or a compatible IP phone or application like Windows Messnger. Support G.711A, G.711?, G.723.1 5.3/6.3 kbps and G.729A/B audio codec. LCD display is available to configure the Phone, besides the Web Based configuration of the same. Electric requirements: ? Voltage: 7.5V 9V DC ? Power: 5.5W 6.5W (max.) ? Power adapter: AC/DC input110 - 240V ? Network intrface:1/2 RJ45 Ethernet connectors. Dimensions: 210×170×60 mm (L × W × H) Price: $75.00 + Shipping for the first 200 responders. And $99.99 + shipping after that. Please send your responses to : [EMAIL PROTECTED] so that someone will call you back. Netweb Group, Inc., 385 Main Street, Suite # 4A, Metuchen, NJ, 08840: Ph: 732-213-2422 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Thursday, July 29, 2004 12:01 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] One More IP Phone for interoperability with Asterisk Holger Schurig wrote: This is phone and the ATA is available soon from http://www.eezeephone.com priced at $75.00 each. This is one more phone based on the PA168 chipset. I guess they're all compatible with Asterisk. I recently added the pages Atron AND PA168 to the wiki. How did you work that out? All the IP phones I can see there are video phones. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and festival
On Thu, 2004-07-29 at 11:32, Adam Lewis wrote: I setup and extension to test festival and when I dial it I get __SNIP__ -- Executing Answer(SIP/phone4-17ae, ) in new stack -- Executing Festival(SIP/phone4-17ae, mary had a little lamb) in new stack == Parsing '/etc/asterisk/festival.conf': == Parsing '/etc/asterisk/festival.conf': Found telco-pbx*CLI SIOD ERROR: unbound variable : tts_textasterisk Jul 29 10:59:08 WARNING[1015826]: app_festival.c:440 festival_exec: Festival returned ER == Spawn extension (sip, 555, 2) exited non-zero on 'SIP/phone4-17ae' __SNIP__ Did you also apply the /usr/src/asterisk/contrib/festival-1.4.3.diff patch? -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] incoming caller doesn't hear rining.
That's true, but in my dialplan, unless a user answers, the call passes on to voicemail. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andy Powell Sent: Thursday, July 29, 2004 5:06 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] incoming caller doesn't hear rining. On 29/07/2004 at 15:49 Johan wrote: A very helpful person just sorted the problem out. Apparently, changing the incoming dial in extensions.conf to Tr solved the problem. Thanks ...but your caller will get a ringing tone even if your phone number is engaged... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
By the way has anyone contacted / invited Stephen Wingfield of Bicom Systems or found (Bout (Bwhether they are going to participate in the conference. (B (BHave anyone of you saw a demo of their GUI Software for Asterisk Management or at (Bleast visited their site. (BBicom systems has a pretty good tool. You must see this. (I may soon be one of their (Bfirst buyers of this high productivity tool.) (B (BThe url for the site is http://www.bicomsystems.com (B (BSeshu Kanuri (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where to start asterisk sourcecode
ShanKutti [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) I would like to study the asterisk source code(Program). I dont' know from which file i've to start reading the code. can anyone helpme. One copy is enough. Two could be an accident. Three, posted once per hour, is excessive. Are you planning upon posting the exact same question again? Patience is a virtue. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
