Re: [asterisk-users] Open CallerID Database?
On Mon, Feb 19, 2007 at 09:02:56PM -0500, C F wrote: I doubt it's CNAM since it has old an outdated listings. On 2/19/07, Paul [EMAIL PROTECTED] wrote: Does google really have the true CNAM database? When I enter my number, I get a search result for my business listing at yellowpages.com Are you referring to something available in a google area other than the search engine? Well, at one time Google got a large telno to name database. I don't know if they have updated it. They can certainly afford to. There are other web sites that do reverse number lookups as well. Still, starting with their database seems a good choice. They might not like you scraping it at once but a thousand * boxes pulling records one call at a time is not something they are going to be bothered by. If this, combined with other info from other sources (including contributions from people who have CNAM) builds a workable database, you will eventually get the LECs contributing their data to it. People want their name to show up correctly. If millions start using a database, the LECs will want their data in it, especially if entry is free or near free for bulk entries. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Communication between servers
This is my problem:
I have two server,
server 1
in iax.conf
[general]
register= isdn_server:[EMAIL PROTECTED]
[voip_server]
type=friend
username=damiano_voip
auth=md5
host=dynamic
secret=voip
qualify=yes
in extensions.conf
exten=_50XXX,1,Dial(IAX2/voip_server/${EXTEN:2},30)
server 2
in iax.conf
[general]
register= voip_server:[EMAIL PROTECTED]
[isdn_server]
type=friend
user=damiano_isdn
auth=md5
host=dynamic
secret=isdn
qualify=yes
in extensions.conf
exten=_45XXX,1,Dial(IAX2/isdn_server/${EXTEN:2},30)
I do correctly calls between the two server, but in the server 1 CLI I see
this:
(calling from a phone registered with server 2 to a phone registered with
server 1)
-- Accepting UNAUTHENTICATED call from 169.254.68.200:
requested format = gsm,
requested prefs = (gsm),
actual format = gsm,
host prefs = (gsm),
priority = mine
-- Executing ChanIsAvail(IAX2/voip_server-14,
IAX2/202SIP/202mISDN/1/202) in new stack
-- Hungup 'IAX2/202-16'
-- Executing Dial(IAX2/voip_server-14, IAX2/202|30) in new stack
-- Called 202
-- Call accepted by 169.254.68.121 (format gsm)
-- Format for call is gsm
-- IAX2/202-19 is ringing
-- Hungup 'IAX2/202-19'
Why do I see UNAUTHENTICATED call?
while in the sever 2 CLI I see:
-- Accepting AUTHENTICATED call from 169.254.68.121 :
requested format = gsm,
requested prefs = (),
actual format = gsm,
host prefs = (gsm),
priority = mine
-- Executing Dial(IAX2/200-3, IAX2/isdn_server/202|30) in new stack
-- Called isdn_server/202
-- Call accepted by 169.254.68.251 (format gsm)
-- Format for call is gsm
-- IAX2/isdn_server-6 is ringing
If I type iax2 show registry in the server 2 CLI I see:
prova*CLI iax2 show registry
Host UsernamePerceived Refresh State
169.254.68.251:4569 voip_serve 169.254.68.200:456960 Registered
and in the server 1 CLI I see:
prova*CLI iax2 show registry
Host UsernamePerceived Refresh State
169.254.68.200:4569 isdn_serve 169.254.68.251:456960 Registered
Thanks for your help, Damiano.
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Re: [asterisk-users] sip to sip ?
why do i need to setup a trunk ? all i want to do is place a sip connection to a remote sip user.. e.g... [EMAIL PROTECTED] On 2/20/07, Mochamad Susantok [EMAIL PROTECTED] wrote: create user trunk on each box and dialplan to make call hi all i've just setup an * box and want to test voip calling, initially from sip user to sip user... local sip users can call each other, no issues. problem arises when i try and call a remote sip account, my * box always returns SIP/2.0 404 Not Found any ideas ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
Hey Guys, I'm glad to see this ignited some discussion. I definitely understand there's some legal implications involved, both on a privacy level, and fraud prevention. Obviously an end-user (ie: the person controlling a listing) has to consent to some sort of release resolving the privacy concerns. I'm somewhat aware of the legal implications involved with storing such personally identifiable information (or whatever the legal term is) and have a concern in making sure such issues are resolved. In reality, how is it efficient for every provider to be running their own database? In my mind, this leaves the horribly evident inaccuracies, and even efficiency issues. Thank God these accuracies aren't integral to the operations of telephony systems. I do understand there is a price to pay for such infrastructure, and I believe that it's obvious the telephony world is riddled with racketeering, price gouging ventures, including companies that charge nearly a $0.01 for a lookup. I realize the following analogy is poor, but in mind this is as close as a internet search engine charging for a basic search query. Infact a basic internet query is much more complex, much more costly (ie: the infrastructure of said systems), and yet self-subsidizing. And to the poster who suggested that I was implying scrapping the results from 411.com, this is definitely not even a remote idea in my mind at all. The basis for my idea was a open, moderated, database that was user controlled and self-subsidized. I know this is way off topic, but I really feel that the telecom industry as a whole, and I'm sure I'm not the only one with this belief, is horribly bloated, running on business models that are clearly 30 years outdated. It is 2007, and with the help of the internet, the exchange of information, these telcos now have real, global competition, and real issues to deal with. Anyways guys, I'm curious to hear your thoughts. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor RingBack Tone Issue
Hi Jean-Marc, I tried to use mixmonitor and seems that it works good. My problem is about calls after a transfer: it seems that asterisk can completely record a call in one file, only in case of blind transfer. If I make an attended transfer I have 2 or more sound files which are impossible to join. Have you successfully recorded sound files of transfered calls in one file?? TIA Giorgio Incantalupo Jean-Marc Salsa wrote: Indeed, perfect ! Thanks a lot ... JM On 2/17/07, *Trevor Peirce* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Jean-Marc Salsa wrote: exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r mailto: SIP/[EMAIL PROTECTED] mailto:SIP/[EMAIL PROTECTED],30,r) Everything works perfectly, except when the softswitch, or the PSTN sends back RingBack Tone. I can see the RTP flow arriving to Asterisk, but, it seems that Asterisk doesn't forward it to the other party (next-hop). Yes because you have the r in there, asterisk sends its own ringing. If you want ringing to be heard from the PSTN, you need to leave that option disabled. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UTStarcom F1000 - WLAN connection unreliable
On Mon, 19 Feb 2007, Anselm Martin Hoffmeister wrote: Does anyone know of those problems, and possibly have a solution? Or just a good idea? I have one of these devices and see the same thing. It works for some time (usually several hours) after registering, after a reboot, then just stops... Requiring a reboot. I use it for a bit of a quick demo of the technology, but I do not think they are reliable enough for production use which is a great shame as they are neat little devices. A friend has similar problems with the clamshell version too. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] UTStarcom F1000 - WLAN connection unreliable
Hello, I've been working on these phones for more than 6 month, I have exactly the same topology and same issues. I met the guys from UTstarcom, we are currently working with them to try to solve the issues. I'm waiting a new release for F1000 (do you have F1000 or F1000 G?) I also try the F3000, I have globally the same issues. (Disconnection, sometimes it reconnects, sometimes no) Do you also have voice quality issues with it or the sound is 'perfect'? Cdt Cyril -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Anselm Martin Hoffmeister Envoyé : lundi 19 février 2007 20:33 À : Asterisk Users Objet : [asterisk-users] UTStarcom F1000 - WLAN connection unreliable Hi list, I bought two UTStarcom F1000 phones, pre-equipped with the latest firmware, including WPA support. Those are configured to register to an asterisk server on the internet (not LAN), and registration works. Calling and being called also, with transfer and all bells and whistles. After a few minutes up to 5 hours (varies widely), the display tells me that an Accesspoint is not available (although it is, with the other phone or a laptop). It will only re-find the WLAN after either powering down the phone, or going into the WLAN settings menu, down to any setting, OK'ing that and activating that WLAN setting. I used any of the profiles 1 to 4 in the meantime, all the same results. I tried changing from WPA to WEP-128 to unencrypted WLAN, IP via DHCP versus static IP, DNS via DHCP (while IP came from DHCP) versus static DNS server, registering to a domain name versus registering to the appropriate IP address - to no avail. I had both phones turned on at times, or only one, that would not make a difference. This occurs with both phones, and on Accesspoints from Buffalo(OpenWRT), Fon (LaFonera), AVM (FritzBoxFon 7050), and T-Com (Eumex something). I did not cross-test all possible combinations - that would be a lot - but quite some. Does anyone know of those problems, and possibly have a solution? Or just a good idea? Is there a known reliable setup? Would anyone care to post what makes his asterisk work with the F1000 (WLAN settings, and sip.conf settings, just to go sure?) Would chances of a working setup increase with asterisk on the LAN (which would make those phones worthless for me...)? My sip.conf relevant parts are [sip505] mailbox=05 callerid=505 type=friend username=sip505 secret=abcd123 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw Thanks for all input, Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with dialout option in voicemail.conf
hello all, i have a small setup in my office which can just send voicemails and retrive them on a LAN now we wanted to go for a nat with the 2 different contexts with entirely different environement the problem i have faced is: when one of the local guy leaves a message i can call him back using his extension as callback property in the voicemail.conf if some outside guy leaves a message means i need to include his context separately using a separate mailboxid and password if the no of users increses and if they are not listed as users in my asterisk box means how can i callback them when i review my voicemails using callback property in voicemail.conf thanks in advance regards asima ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] UTStarcom F1000 - WLAN connection unreliable
A friend has one I helped him set up, and it's not up to production use. Which is sad as they would like to buy more of them. PaulH On Tue, 2007-02-20 at 10:28 +0100, Cyril Mandrilly wrote: Hello, I've been working on these phones for more than 6 month, I have exactly the same topology and same issues. I met the guys from UTstarcom, we are currently working with them to try to solve the issues. I'm waiting a new release for F1000 (do you have F1000 or F1000 G?) I also try the F3000, I have globally the same issues. (Disconnection, sometimes it reconnects, sometimes no) Do you also have voice quality issues with it or the sound is 'perfect'? Cdt Cyril -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Anselm Martin Hoffmeister Envoyé : lundi 19 février 2007 20:33 À : Asterisk Users Objet : [asterisk-users] UTStarcom F1000 - WLAN connection unreliable Hi list, I bought two UTStarcom F1000 phones, pre-equipped with the latest firmware, including WPA support. Those are configured to register to an asterisk server on the internet (not LAN), and registration works. Calling and being called also, with transfer and all bells and whistles. After a few minutes up to 5 hours (varies widely), the display tells me that an Accesspoint is not available (although it is, with the other phone or a laptop). It will only re-find the WLAN after either powering down the phone, or going into the WLAN settings menu, down to any setting, OK'ing that and activating that WLAN setting. I used any of the profiles 1 to 4 in the meantime, all the same results. I tried changing from WPA to WEP-128 to unencrypted WLAN, IP via DHCP versus static IP, DNS via DHCP (while IP came from DHCP) versus static DNS server, registering to a domain name versus registering to the appropriate IP address - to no avail. I had both phones turned on at times, or only one, that would not make a difference. This occurs with both phones, and on Accesspoints from Buffalo(OpenWRT), Fon (LaFonera), AVM (FritzBoxFon 7050), and T-Com (Eumex something). I did not cross-test all possible combinations - that would be a lot - but quite some. Does anyone know of those problems, and possibly have a solution? Or just a good idea? Is there a known reliable setup? Would anyone care to post what makes his asterisk work with the F1000 (WLAN settings, and sip.conf settings, just to go sure?) Would chances of a working setup increase with asterisk on the LAN (which would make those phones worthless for me...)? My sip.conf relevant parts are [sip505] mailbox=05 callerid=505 type=friend username=sip505 secret=abcd123 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw Thanks for all input, Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mask the caller-ID
Dear All : I need to mask the caller ID and pretend to make a transfer call from another extension : exten = 558,1,Answer exten = 558,2,Playback(soundclip) exten = 558,3,Dial(SIP/[EMAIL PROTECTED]) The scenario is like this : Someone is calling 558 at my company - he will hear a soundclip voice message then I will direct it to extension 472 I need 472 to not see the extension of the caller-ID I need him to see it coming from another extension let us say : 111 ... How can I do this ? Mohamed Farid ,, * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This e-mail (including attachments) is classified as Mediterranean Smart Cards Company confidential and proprietary information The recipient hereby is committed to hold in strict confidence the contents of this (e-mail, document, and information) and not to disclose to any third party without the prior written consent of Mediterranean Smart Cards Company. Recipient will be held liable for any unauthorized disclosure. It is intended solely for the addressee. Unless you are the addressee, you may not read, copy, use or store this e-mail in any way, or permit others to. If you have received it in error, please notify the sender by return e-mail and delete the message in its entirety, including any attachments * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip to sip ?
Dennis Kavadas wrote:
hi all
i've just setup an * box and want to test voip calling, initially from
sip user to sip user...
local sip users can call each other, no issues.
problem arises when i try and call a remote sip account, my * box
always returns SIP/2.0 404 Not Found
any ideas ?
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Dennis
I use the following as my default context:-
[default]
exten = _X.,1,NoOp(Incoming Call from ${CALLERID} for
[EMAIL PROTECTED])
exten = _X.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten = _X.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten = _X.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
exten = _X.,5,GotoIf($[${SIPDOMAIN} = ${MYIP}]?10)
exten = _X.,6,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
exten = _X.,7,NoOp(@${SIPDOMAIN} is remote - forwarding...)
exten = _X.,8,Macro(uridial,[EMAIL PROTECTED])
exten = _X.,9,HangUp()
exten = _X.,10,Goto(default-noturi,${EXTEN},1)
exten = h,1,HangUp()
exten = s-BUSY,1,Congestion
exten = s-CHANUNAVAIL,1,Congestion
exten = s-CONGESTION,1,Congestion
[macro-uridial]
exten = s,1,NoOp(Outbound SIP URI call ${ARG1})
exten = s,2,SetCIDNum(0123456789)
exten = s,3,Dial(SIP/${ARG1})
exten = s,4,Congestion()
HTH
--
Chris Hills | Tel: +44 (0)1527 572754
IT Services | Fax: +44 (0)1527 572901
North East Worcestershire College | Web: http://www.ne-worcs.ac.uk/
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[asterisk-users] Agents busy in queue
I need some help with a problem which I'm facing with Asterisk 1.4 final release. I'm using static agents in a queue. Sometimes when an agent answers a call in queue and then releases it, the status for that agent in the queue remains busy where as there is not channel associated to that SIP client. For furthur calls in that queue that particular agent receives no more calls unless you unregister and then register that SIP client. This is occuring very regularly. Any one with a solution or idea?? Thanks, Kashif. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: zaptel 1.4.0 on Fedora Core 6 x86_64
I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on FC5 x86_64 very good, but since FC keeps updating, I tried to follow newer kernel versions. I can't pass the zaptel compilation. Everything is OK, but when I finished, and tried to load it, allways got module not found when I run modprobe zaptel, and modprobe ztdummy. I already tried to modify is with the sed 1 option but doesn't work. I'm running make linux26, make install. Also, I have the kernel sources, and a symlink to /lib/modules/ Also, I tried the make install-udev, since there was no zap device on /dev/zap but nothing. The error is that when I run modprobe the result is FATAL NO ZAPTEL MODULE FOUND. Any clue about this? Thanks Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: zaptel 1.4.0 on Fedora Core 6 x86_64
I forget, today I update my Fedora Core 5 from 2.6.18.1-2257 to 2.6.19-2288, of course I recompile then zaptel. And voila, don't work any more. Now I have both FC5 FC6 with no longer ztdummy working. Do I need to go back to MS-DOS 3.22? Thanks, Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agents busy in queue
Are you using attended transfers? PaulH On Tue, 2007-02-20 at 15:37 +0500, Kashif Anwar wrote: I need some help with a problem which I'm facing with Asterisk 1.4 final release. I'm using static agents in a queue. Sometimes when an agent answers a call in queue and then releases it, the status for that agent in the queue remains busy where as there is not channel associated to that SIP client. For furthur calls in that queue that particular agent receives no more calls unless you unregister and then register that SIP client. This is occuring very regularly. Any one with a solution or idea?? Thanks, Kashif. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
CA == Carlos Alperin [EMAIL PROTECTED] writes: CA The error is that when I run modprobe the result is FATAL NO CA ZAPTEL MODULE FOUND. CA Any clue about this? It is important that you do not rephrase error messages, but copy them directly. I probably can't help you even with the correct information though. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] UTStarcom F1000 - WLAN connection unreliable
On Tue, 20 Feb 2007, Cyril Mandrilly wrote: Hello, I've been working on these phones for more than 6 month, I have exactly the same topology and same issues. I met the guys from UTstarcom, we are currently working with them to try to solve the issues. I'm waiting a new release for F1000 (do you have F1000 or F1000 G?) I also try the F3000, I have globally the same issues. (Disconnection, sometimes it reconnects, sometimes no) Do you also have voice quality issues with it or the sound is 'perfect'? For me, I have the F1000G. Voice quality is good - only tried uLaw codec, so I'd expect it to be as good as anything else. I'd love to recomend these to clients (they even have a neat little desk stand/charger, standby + talk time is fantastic), but the frustration at keeping synced is, er, frustrating! Gordon Cdt Cyril -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Anselm Martin Hoffmeister Envoyé : lundi 19 février 2007 20:33 À : Asterisk Users Objet : [asterisk-users] UTStarcom F1000 - WLAN connection unreliable Hi list, I bought two UTStarcom F1000 phones, pre-equipped with the latest firmware, including WPA support. Those are configured to register to an asterisk server on the internet (not LAN), and registration works. Calling and being called also, with transfer and all bells and whistles. After a few minutes up to 5 hours (varies widely), the display tells me that an Accesspoint is not available (although it is, with the other phone or a laptop). It will only re-find the WLAN after either powering down the phone, or going into the WLAN settings menu, down to any setting, OK'ing that and activating that WLAN setting. I used any of the profiles 1 to 4 in the meantime, all the same results. I tried changing from WPA to WEP-128 to unencrypted WLAN, IP via DHCP versus static IP, DNS via DHCP (while IP came from DHCP) versus static DNS server, registering to a domain name versus registering to the appropriate IP address - to no avail. I had both phones turned on at times, or only one, that would not make a difference. This occurs with both phones, and on Accesspoints from Buffalo(OpenWRT), Fon (LaFonera), AVM (FritzBoxFon 7050), and T-Com (Eumex something). I did not cross-test all possible combinations - that would be a lot - but quite some. Does anyone know of those problems, and possibly have a solution? Or just a good idea? Is there a known reliable setup? Would anyone care to post what makes his asterisk work with the F1000 (WLAN settings, and sip.conf settings, just to go sure?) Would chances of a working setup increase with asterisk on the LAN (which would make those phones worthless for me...)? My sip.conf relevant parts are [sip505] mailbox=05 callerid=505 type=friend username=sip505 secret=abcd123 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw Thanks for all input, Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-1.2.10 not releasing SIP sessions
Hi, It's really weired issue,I'm facing with asterisk-1.2.10 version. I see SIP call sessions stuck in asterisk for hours and then somehow get released. There happens to be an issue with BYE/CANCEL release msgs b/w sip entities. Has anyone faced this issue before also rtptimeout option given in sip.conf is not helping out. Any suggestions? -AG x post to *-dev, *-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-1.2.10 not releasing SIP sessions
ast guy wrote: Hi, It's really weired issue,I'm facing with asterisk-1.2.10 version. I Upgrade to 1.2.15 and see if it's still an issue. Also, cross posting isn't suggested. Many on the Dev list are also subscribed to User. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP interface status and calllimit
James Fromm wrote: There is an issue when using call-limit for a SIP interface in sip.conf. The call count does not properly reset when some calls end. The problem happens regardless of which side of the connection ends the call. It happens on all calls including calls from SIP interface to SIP interface (with no reinvite) within the same Asterisk server. I have not been able to determine a definite pattern. I can call from one interface to another 50 times before it happens and sometimes it happens after only 2 calls. We have to enable call-limit for our customer service queue agents so that the ringinuse option in queues.conf will work properly. Has anyone else seen this issue? Any ideas? This doesn't really help you, but might help others when deciding how to design their Asterisk system. On our phones we set call waiting off and each line appearance registers as a separate SIP user. This avoids all this silliness with call limits, group limits, etc. This also allows us total control about which call appearance a call shows up on, roll over and hunting features, etc. It does require a little more work in the dialplan, but for our needs it is well worth it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: zaptel 1.4.0 on Fedora Core 6 x86_64
On Tue, 2007-02-20 at 06:01 -0500, Carlos Alperin wrote: I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on FC5 x86_64 very good, but since FC keeps updating, I tried to follow newer kernel versions. I build an RPM on i386 Fedora 7 Test 1 updated to Rawhide and it seems to work ok too. I can't pass the zaptel compilation. Everything is OK, but when I finished, and tried to load it, allways got module not found when I run modprobe zaptel, and modprobe ztdummy. If the modules aren't there how can you say everything is ok? :) Clearly something must be going wrong. I already tried to modify is with the sed 1 option but doesn't work. What did you modify with sed? I'm running make linux26, make install. Also, I have the kernel sources, and a symlink to /lib/modules/ If I remember correctly you need to specify make all. If you had paid any attention to the *entire* output of the compilation process than you would have noticed an error about this. Yes there is a lot of output but you really need to keep an eye on it. See if this helps: $ make clean $ make menuselect $ make all $ make install Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] unwanted chanspy: strange behaviour
Hi, I have an Asterisk box with a Sangoma PRI on a Debian distro. It may happen that user A tries to call B but for some reason the call drops and A is connected to another call between C and D. User A can only hear what C and D are saying, like a sort of unwanted chanspy. Is there anybody who experienced a strange behaviour like this? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: zaptel 1.4.0 on Fedora Core 6 x86_64
On Tue, 2007-02-20 at 06:02 -0500, Carlos Alperin wrote: I forget, today I update my Fedora Core 5 from 2.6.18.1-2257 to 2.6.19-2288, of course I recompile then zaptel. And voila, don't work any more. Now I have both FC5 FC6 with no longer ztdummy working. I updated and don't work anymore is hardly any detailed information of what you did so we may be able to help you. Or do you think we can guess what you did, what happened and why it went wrong? Do I need to go back to MS-DOS 3.22? Whatever floats your boat. Please don't take your frustration out on the community. We are not the cause of your problems. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip to sip ?
If you're getting a 404, I would assume it is reacting like any other
non-connection would (http, etc). Do you know if the packets are
reaching the phone, or if the phone is registering its correct IP
Address? If it is registering, but no packets are reaching it, could it
be a routing issue?
Rob
Chris Hills wrote:
Dennis Kavadas wrote:
hi all
i've just setup an * box and want to test voip calling, initially from
sip user to sip user...
local sip users can call each other, no issues.
problem arises when i try and call a remote sip account, my * box
always returns SIP/2.0 404 Not Found
any ideas ?
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Dennis
I use the following as my default context:-
[default]
exten = _X.,1,NoOp(Incoming Call from ${CALLERID} for
[EMAIL PROTECTED])
exten = _X.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten = _X.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten = _X.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
exten = _X.,5,GotoIf($[${SIPDOMAIN} = ${MYIP}]?10)
exten = _X.,6,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
exten = _X.,7,NoOp(@${SIPDOMAIN} is remote - forwarding...)
exten = _X.,8,Macro(uridial,[EMAIL PROTECTED])
exten = _X.,9,HangUp()
exten = _X.,10,Goto(default-noturi,${EXTEN},1)
exten = h,1,HangUp()
exten = s-BUSY,1,Congestion
exten = s-CHANUNAVAIL,1,Congestion
exten = s-CONGESTION,1,Congestion
[macro-uridial]
exten = s,1,NoOp(Outbound SIP URI call ${ARG1})
exten = s,2,SetCIDNum(0123456789)
exten = s,3,Dial(SIP/${ARG1})
exten = s,4,Congestion()
HTH
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Re: [asterisk-users] SIP interface status and calllimit
We do the same thing only we use ringinuse=no and autopause=yes for the queue. With autopause, if the agent is busy their interface in the queue gets paused. Setting call-limit for the SIP interface is the only way to make ringinuse=no work. Eric ManxPower Wieling wrote: James Fromm wrote: There is an issue when using call-limit for a SIP interface in sip.conf. The call count does not properly reset when some calls end. The problem happens regardless of which side of the connection ends the call. It happens on all calls including calls from SIP interface to SIP interface (with no reinvite) within the same Asterisk server. I have not been able to determine a definite pattern. I can call from one interface to another 50 times before it happens and sometimes it happens after only 2 calls. We have to enable call-limit for our customer service queue agents so that the ringinuse option in queues.conf will work properly. Has anyone else seen this issue? Any ideas? This doesn't really help you, but might help others when deciding how to design their Asterisk system. On our phones we set call waiting off and each line appearance registers as a separate SIP user. This avoids all this silliness with call limits, group limits, etc. This also allows us total control about which call appearance a call shows up on, roll over and hunting features, etc. It does require a little more work in the dialplan, but for our needs it is well worth it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: FW: [asterisk-users] Problem Transferring Direct to Voicemail
Try version 1.2.6. On 2/16/07, Savoy, Kevin - Williston, ND [EMAIL PROTECTED] wrote: Well thanks to those who did reply. I guess I'll have to live with it until somehow it gets fixed. The reason I upgraded to 1.4 is that there were three or four other issues I had that this fixed. Going back just isn't really an option since those issues were bigger then this one. Guess we'll live with it for now. If anyone ever hears of this and a fix for it please let me know. Again thanks for responding this time. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Friday, February 16, 2007 2:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: FW: [asterisk-users] Problem Transferring Direct to Voicemail Maybe nobody knows. I certainty know that I've never ever seen that error. Savoy, Kevin - Williston, ND wrote: Could someone at least respond to this so that I know it is getting out there? I have posted this three times and not gotten one single response. I even totally reworded it hoping that would help. I'm at a loss here and not sure where to turn next. All searches I've done come up with nothing telling me what Notify answer on an owned channel means and what to do about it. PLEASE!! Someone?? Anyone??? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Kernel and zaptel versions
If your connections are VoIP, the first area to look at for quality is network jitter/congestion/drops. I'm mostly worried about drops. A little bit of garbling I can deal with but a dropped call is just VERY bad. Especially when it happens again and again. Does anyone know any methods for tracing dropped calls? All I see is a normal hangup in the logs. The dropped calls seem VERY random and happen regardless of VSP. All I can determine is that it's asterisk that's at fault but I really have no justification for it. All of our calls are VOIP only. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 bit HPEC modules available?
Tony Nichols wrote: I talked to tech support today... no 64bit yet. We received the 64-bit modules yesterday and they are undergoing testing as I write this. If they pass inspection, they'll be placed on the FTP site in the next couple of days and an announcement will be made. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto load of zap drivers
On Tue, Feb 20, 2007 at 04:45:34PM +1100, Klaverstyn, David C wrote: I am running CentOS 4.4. You say I need modprobe ztdummy on startup. I though the udev option made that happen. No. look for 'modprobe ztdummy' in the zaptel init.d script. If this is all you need, you can trim out much of it. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
FATAL: Module zaptel not found. That is the message -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benny Amorsen Sent: Tuesday, February 20, 2007 6:32 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64 CA == Carlos Alperin [EMAIL PROTECTED] writes: CA The error is that when I run modprobe the result is FATAL NO ZAPTEL CA MODULE FOUND. CA Any clue about this? It is important that you do not rephrase error messages, but copy them directly. I probably can't help you even with the correct information though. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP interface status and calllimit
It does. Eric ManxPower Wieling wrote: Maybe Queue doesn't consider a SIP account that returns BUSY as in use. That would be the only case where I could see needing call-limit. James Fromm wrote: We do the same thing only we use ringinuse=no and autopause=yes for the queue. With autopause, if the agent is busy their interface in the queue gets paused. Setting call-limit for the SIP interface is the only way to make ringinuse=no work. Eric ManxPower Wieling wrote: James Fromm wrote: There is an issue when using call-limit for a SIP interface in sip.conf. The call count does not properly reset when some calls end. The problem happens regardless of which side of the connection ends the call. It happens on all calls including calls from SIP interface to SIP interface (with no reinvite) within the same Asterisk server. I have not been able to determine a definite pattern. I can call from one interface to another 50 times before it happens and sometimes it happens after only 2 calls. We have to enable call-limit for our customer service queue agents so that the ringinuse option in queues.conf will work properly. Has anyone else seen this issue? Any ideas? This doesn't really help you, but might help others when deciding how to design their Asterisk system. On our phones we set call waiting off and each line appearance registers as a separate SIP user. This avoids all this silliness with call limits, group limits, etc. This also allows us total control about which call appearance a call shows up on, roll over and hunting features, etc. It does require a little more work in the dialplan, but for our needs it is well worth it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] FW: zaptel 1.4.0 on Fedora Core 6 x86_64
Ok, let 's go one at the time: 1. I didn't build any rpm. I just install zaptel, libpri asterisk 1.4 2. I can run asterisk, that is what I mean that everything goes OK, No ERRORs on compilation. Asterisk works. However, if I try to run zaptel/ztdummy, I cannot since modprobe zaptel reports FATAL: Module zaptel not found. 3. I modified since I saw a couple of recommendations about this problems, and several recommends to modified the ztdummy.c 4. Regarding the procedure: I did make clean, make menuselect, make, make install make config. First time I read about make all. That's going to be my next try, and send the compilation output to a file. Thanks, Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] FW: zaptel 1.4.0 on Fedora Core 6 x86_64
No, I don't blame the community. In the best case, I can blame Fedora community for the non-documented changes. This is only to share experiences that can teach the rest, as I learn reading all the e-mail with other people problem. As I said, I updated the kernel and the recompile zaptel. Then, as in FC6 zaptel ztdummy modules are not longer there. Error in modprobbe: FATAL: Module zaptel not found. On the compilation I couldn't see any error. Now I 'm going to follow your suggestion and output the compilation on a file to search errors. Regards, Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
On Tue, Feb 20, 2007 at 10:34:02AM -0500, Carlos Alperin wrote: FATAL: Module zaptel not found. Was it indeed installed? find /lib/modules/`uname -r` -name zaptel.ko I expect it wasn't, so this should probably not return anything. What happens when you run 'make modues' in the zaptel source directory? What is your kernel version? uname -r Do you have kernel source/headers for that kernel? e.g: ls -l /lib/modules/`uname -r`/build -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
To Patrick Benny all. The make all made the difference in FC5 with kernel 2.6.19-1.2288.fc5 (hardware is x86_64 on AMD Dual Athlon 3800, but I don't think that makes any difference). There was no errors on compilation, but this time it founds the modules, and modprobe works. Thanks, I already stop looking my MS-DOS floppies. Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
Since I found the problem in FC5, I'm going to answer on FC6 that is where still I have the same problem: find /lib/modules/`uname -r` -name zaptel.ko reports nothing. So, I assume that you're right. Kernel Version: 2.6.19-1.2911.fc6xen Yes, I have sources/headers [EMAIL PROTECTED] zaptel-1.4.0]# ls -l /lib/modules/`uname -r`/build lrwxrwxrwx 1 root root 50 Feb 19 13:22 /lib/modules/2.6.19-1.2911.fc6xen/build - ../../../usr/src/kernels/2.6.19-1.2911.fc6-x86_64/ And on the output of make all make install I got no errors reported. Now, only on FC6 I cannot load zaptel ztdummy. Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Inbound Problem
Am working with Arun on this project - here's a longer description of the problem: We've been fighting with our service provider on this issue - we seem to be getting a BYE just after we receive an ACK. They claim that it is an asterisk issue! The service provider provides only IP based authentication for inbound. We have used username-password based authentication with the same setup with *no problems* whatsoever! If we configure an Audiocodes MEdia gateway to receive the calls, there is no issue - so there's something that asterisk is doing? or asterisk-Provider gateway combo? In our efforts to mask IP, I have used PROVIDER-IP for the IP of my service provider (host) and AsteriskIP to indicate my asterisk server sip.conf [PROVIDER] type=peer disallow=all allow=g729 context=default host= fromuser=y.y.y.y port=5060 insecure=very canreinvite=no nat=yes qualify=yes CLI output: -- Executing Answer(SIP/PROVIDER-IP-b7a076a8, ) in new stack We're at 124.7.195.102 port 47698 Adding codec 0x100 (g729) to SDP Reliably Transmitting (NAT) to PROVIDER-IP:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bK6bd3121243ee9f936c4aeb96d6785b7a;received=PROVIDER-IP From: sip:[EMAIL PROTECTED];tag=3380976385-794612 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:8009422419@'AsteriskIP' Content-Type: application/sdp Content-Length: 183 v=0 o=root 2172 2172 IN IP4 AsteriskIP s=session c=IN IP4 AsteriskIP t=0 0 m=audio 47698 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - --- -- Executing Playback(SIP/PROVIDER-IP-b7a076a8, park) in new stack -- Playing 'park' (language 'en') AstSQL*CLI -- SIP read from PROVIDER-IP:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 5 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 From: sip:[EMAIL PROTECTED];tag=3380976385-794612 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Via: SIP/2.0/UDP 221.135.102.100:5060 ;branch=z9hG4bK02505a1dcc5937d9a648eebc0052b422 Content-Length: 0 --- (9 headers 0 lines) --- AstSQL*CLI -- SIP read from PROVIDER-IP:5060: BYE sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 5 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 From: sip:[EMAIL PROTECTED];tag=3380976385-794612 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE Via: SIP/2.0/UDP 221.135.102.100:5060 ;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f Content-Length: 0 --- (9 headers 0 lines) --- Sending to PROVIDER-IP : 5060 (NAT) Transmitting (NAT) to PROVIDER-IP:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f;received=PROVIDER-IP From: sip:[EMAIL PROTECTED];tag=3380976385-794612 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 The following is an ngrep of the traffic for an inbound call - 'U' marks the begin of the packet grabbed. U PROVIDER-IP:5060 - AsteriskIP:5060 INVITE sip:800942@AsteriskIP SIP/2.0..Max-Forwards: 5..Session-Expires: 3600;Refresher=uac..Suppor ted: timer..To: sip:[EMAIL PROTECTED]:5060..From: sip:PROVIDER-IP;tag=3380960452-790279..Co ntact: sip:PROVIDER-IP:5060..Remote-Party-Id: sip:PROVIDER-IP;party=calling;screen=no;privacy =off..Call-ID: [EMAIL PROTECTED]: 1 INVITE..Via: SIP/2.0/UDP 221. 135.102.100:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4..Allow-Events: telephone-event..Content-T ype: application/sdp..Content-Length: 206v=0..o=nextone-msw1 1774 4816 IN IP4 PROVIDER-IP..s=sip call..c=IN IP4 PROV-IP-2..t=0 0..m=audio 18932 RTP/AVP 18 19..a=ptime:20..a=rtpmap:19 CN/8000..a=fm tp:18 annexb=yes..a=rtpmap:18 G729/8000.. # U AsteriskIP:5060 - PROVIDER-IP:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4; received=PROVIDER-IP..From: sip:PROVIDER-IP;tag=3380960452-790279..To: sip:[EMAIL PROTECTED] 11.2:5060..Call-ID: [EMAIL PROTECTED]: 1 INVITE..User-Agent: Ast erisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:[EMAIL PROTECTED]..Content-Length: 0 # U AsteriskIP:5060 - PROVIDER-IP:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4 ;received=PROVIDER-IP..From: sip:PROVIDER-IP;tag=3380960452-790279..To: sip:[EMAIL PROTECTED]:5060;tag=as78bcde29..Call-ID: [EMAIL PROTECTED]: 1 INVITE. .User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:800942@AsteriskIP..Content-Length: 0 # U
Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
On Tue, Feb 20, 2007 at 11:13:34AM -0500, Carlos Alperin wrote: Since I found the problem in FC5, I'm going to answer on FC6 that is where still I have the same problem: find /lib/modules/`uname -r` -name zaptel.ko reports nothing. So, I assume that you're right. Kernel Version: 2.6.19-1.2911.fc6xen Yes, I have sources/headers What exactly? rpm -qa | grep kernel -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
On Tue, Feb 20, 2007 at 11:13:34AM -0500, Carlos Alperin wrote: Since I found the problem in FC5, I'm going to answer on FC6 that is where still I have the same problem: find /lib/modules/`uname -r` -name zaptel.ko reports nothing. So, I assume that you're right. Kernel Version: 2.6.19-1.2911.fc6xen Yes, I have sources/headers [EMAIL PROTECTED] zaptel-1.4.0]# ls -l /lib/modules/`uname -r`/build lrwxrwxrwx 1 root root 50 Feb 19 13:22 /lib/modules/2.6.19-1.2911.fc6xen/build - ../../../usr/src/kernels/2.6.19-1.2911.fc6-x86_64/ And on the output of make all make install I got no errors reported. Hmm... I must have missed those. Going over all the messages in the thread I can see some error reports from modprobe, but not those from 'make' or from 'make install'. In the zaptel build dir: ls -l *.ko -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
Tzafrir, [EMAIL PROTECTED] zaptel-1.4.0]# ls -l *.ko -rw-r--r-- 1 root root 677908 Feb 20 11:06 zaptel.ko -rw-r--r-- 1 root root 187731 Feb 20 11:06 ztd-loc.ko -rw-r--r-- 1 root root 163064 Feb 20 11:06 ztdummy.ko -rw-r--r-- 1 root root 173618 Feb 20 11:06 zttranscode.ko And those are the modules that I choose on make menuselect. Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, February 20, 2007 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64 On Tue, Feb 20, 2007 at 11:13:34AM -0500, Carlos Alperin wrote: Since I found the problem in FC5, I'm going to answer on FC6 that is where still I have the same problem: find /lib/modules/`uname -r` -name zaptel.ko reports nothing. So, I assume that you're right. Kernel Version: 2.6.19-1.2911.fc6xen Yes, I have sources/headers [EMAIL PROTECTED] zaptel-1.4.0]# ls -l /lib/modules/`uname -r`/build lrwxrwxrwx 1 root root 50 Feb 19 13:22 /lib/modules/2.6.19-1.2911.fc6xen/build - ../../../usr/src/kernels/2.6.19-1.2911.fc6-x86_64/ And on the output of make all make install I got no errors reported. Hmm... I must have missed those. Going over all the messages in the thread I can see some error reports from modprobe, but not those from 'make' or from 'make install'. In the zaptel build dir: ls -l *.ko -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk / ACT CRM Integration
Has anyone ever been party to an integration of ACT CRM platform with Asterisk? Thanks Cory Andrews ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
On Tue, Feb 20, 2007 at 11:59:56AM -0500, Carlos Alperin wrote: Tzafrir, [EMAIL PROTECTED] zaptel-1.4.0]# ls -l *.ko -rw-r--r-- 1 root root 677908 Feb 20 11:06 zaptel.ko -rw-r--r-- 1 root root 187731 Feb 20 11:06 ztd-loc.ko -rw-r--r-- 1 root root 163064 Feb 20 11:06 ztdummy.ko -rw-r--r-- 1 root root 173618 Feb 20 11:06 zttranscode.ko And those are the modules that I choose on make menuselect. So what is the output of: make install -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] analog channels calling out not detect DTMF
I have a TDM2402E card. Occasionally I have noticed that a number I call that gives and IVR the DTMF keys are not detected. All other times the DTMF works fine. /proc/interrupts is: CPU0 CPU1 0: 500798020 500749281IO-APIC-edge timer 8: 4 9IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 15:45059124502264IO-APIC-edge ide1 177:16580511649817 IO-APIC-level libata 185:2919898 0 IO-APIC-level eth1 193: 500724461 500697804 IO-APIC-level wctdm24xxp 201: 192176762 IO-APIC-level eth0 NMI: 0 0 LOC: 1001615317 1001615316 ERR: 0 MIS: 0 What might I look to getting this to work all the time. Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
Tzafrir,
This is the output of make install:
make[1]: Entering directory `/usr/src/zaptel-1.4.0'
make -C /lib/modules/2.6.19-1.2911.fc6xen/build
SUBDIRS=/usr/src/zaptel-1.4.0 modules
make[2]: Entering directory `/usr/src/kernels/2.6.19-1.2911.fc6-x86_64'
Building modules, stage 2.
MODPOST 4 modules
make[2]: Leaving directory `/usr/src/kernels/2.6.19-1.2911.fc6-x86_64'
make[1]: Leaving directory `/usr/src/zaptel-1.4.0'
build_tools/genudevrules /etc/udev/rules.d/zaptel.rules
if [ -d /usr/lib/hotplug/firmware ]; then \
/usr/bin/install -c -m 644 wct4xxp/*.ima
/usr/lib/hotplug/firmware; \
fi
if [ -d /lib/firmware ]; then \
/usr/bin/install -c -m 644 wct4xxp/*.ima /lib/firmware; \
fi
Installed firmware
/usr/bin/install -c -D -m 755 libtonezone.a /usr/lib/libtonezone.a
/usr/bin/install -c -D -m 755 libtonezone.so /usr/lib/libtonezone.so.1.0
if [ -z -a `id -u` = 0 ]; then \
/sbin/ldconfig || : ;\
fi
rm -f /usr/liblibtonezone.so
/bin/ln -sf libtonezone.so.1.0 \
/usr/lib/libtonezone.so.1
/bin/ln -sf libtonezone.so.1.0 \
/usr/lib/libtonezone.so
if [ -z ] [ -x /usr/sbin/sestatus ] (/usr/sbin/sestatus | grep
SELinux
status: | grep -q enabled) ; then restorecon -v /usr/lib/libtonezone.so;
fi
/usr/bin/install -c -D -m 644 zaptel.h /usr/include/zaptel/zaptel.h
/usr/bin/install -c -D -m 644 tonezone.h /usr/include/zaptel/tonezone.h
rm -f /usr/include/linux/zaptel.h
rm -f /usr/include/linux/torisa.h
rm -f /usr/include/zaptel.h
rm -f /usr/include/torisa.h
rm -f /usr/include/tonezone.h
if [ -f ztcfg ]; then \
/usr/bin/install -c -D -m 755 ztcfg /sbin/ztcfg; \
fi
if [ -f sethdlc-new ]; then \
/usr/bin/install -c -D -m 755 sethdlc-new /sbin/sethdlc; \
elif [ -f sethdlc ]; then \
/usr/bin/install -c -D -m 755 sethdlc /sbin/sethdlc; \
fi
if [ -f zttool ]; then \
/usr/bin/install -c -D -m 755 zttool /sbin/zttool; \
fi
for x in zaptel.ko ztd-loc.ko ztdummy.ko zttranscode.ko; do \
rm -f /lib/modules/2.6.19-1.2911.fc6xen/extra/$x ; \
done; \
make -C /lib/modules/2.6.19-1.2911.fc6xen/build
SUBDIRS=/usr/src/zaptel-
1.4.0 INSTALL_MOD_PATH= INSTALL_MOD_DIR=misc modules_install; \
if [ -f datamods/syncppp.ko ]; then \
make -C datamods install; \
else \
rm -f
/lib/modules/2.6.19-1.2911.fc6xen/misc/{hdlc_*,syncppp}.ko
; \
fi
make[1]: Entering directory `/usr/src/kernels/2.6.19-1.2911.fc6-x86_64'
INSTALL /usr/src/zaptel-1.4.0/zaptel.ko
INSTALL /usr/src/zaptel-1.4.0/ztd-loc.ko
INSTALL /usr/src/zaptel-1.4.0/ztdummy.ko
INSTALL /usr/src/zaptel-1.4.0/zttranscode.ko
DEPMOD 2.6.19-1.2911.fc6
make[1]: Leaving directory `/usr/src/kernels/2.6.19-1.2911.fc6-x86_64'
if ! [ -f wcfxsusb.o ]; then \
rm -f /lib/modules/2.6.19-1.2911.fc6xen/misc/wcfxsusb.o; \
fi; \
rm -f /lib/modules/2.6.19-1.2911.fc6xen/misc/wcfxs.o
/usr/bin/install -c -m 644 doc/ztcfg.8 /usr/share/man/man8
/usr/bin/install -c -m 644 doc/zttool.8 /usr/share/man/man8
[ `id -u` = 0 ] /sbin/depmod -a 2.6.19-1.2911.fc6xen || :
[ -f /etc/zaptel.conf ] || /usr/bin/install -c -D -m 644 zaptel.conf.sample
/etc
/zaptel.conf
build_tools/genmodconf linux26 ztd-loc ztdummy zttranscode
Building /etc/modprobe.d/zaptel...
***
*** WARNING:
*** If you had custom settings in /etc/modprobe.d/zaptel,
*** they have been moved to /etc/modprobe.d/zaptel.bak.
***
*** In the future, do not edit /etc/modprobe.d/zaptel, but
*** instead put your changes in another file
*** in the same directory so that they will not
*** be overwritten by future Zaptel updates.
***
I cannot see any error.
The Warning is due to I already compiled zaptel several times, and
zaptel.conf, and zaptel already exists.
Carlos Alperin
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RE: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
Tzafrir, Sorry, I didn't see this e-mail before: What exactly? rpm -qa | grep kernel [EMAIL PROTECTED] zaptel-1.4.0]# rpm -qa | grep kernel kernel-xen-2.6.18-1.2798.fc6 kernel-xen-2.6.19-1.2911.fc6 kernel-headers-2.6.19-1.2911.fc6 kernel-devel-2.6.19-1.2911.fc6 kernel-devel-2.6.19-1.2911.fc6 Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] B410P - Please an advise
Dear all, I tried everything to make my Digium B410P card working. I'm using Trixbox 2 and I recompiled Zaptel, Asterisk and did make b410p. Everything goes well but at the end I cannot use the card. I don't think the card is broken as in some trials in the past I saw the red blinking lights. If I put dmesg | grep Digium I cannot see anything and in asterisk cli I don't have misdn command. What happened? I tried to use baronet misdn script. no errors but the same result as above. I tried to compile manually from misdn website and again I got the same result. I tried to use the baronet script that installs everything (zaptel, asterisk, misdn) but no joy. If anyone knows anything else I can try, please let me know. Thanks, Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
On Tue, Feb 20, 2007 at 12:54:15PM -0500, Carlos Alperin wrote:
Tzafrir,
This is the output of make install:
make[1]: Entering directory `/usr/src/zaptel-1.4.0'
make -C /lib/modules/2.6.19-1.2911.fc6xen/build
SUBDIRS=/usr/src/zaptel-1.4.0 modules
make[2]: Entering directory `/usr/src/kernels/2.6.19-1.2911.fc6-x86_64'
Building modules, stage 2.
MODPOST 4 modules
make[2]: Leaving directory `/usr/src/kernels/2.6.19-1.2911.fc6-x86_64'
make[1]: Leaving directory `/usr/src/zaptel-1.4.0'
build_tools/genudevrules /etc/udev/rules.d/zaptel.rules
if [ -d /usr/lib/hotplug/firmware ]; then \
/usr/bin/install -c -m 644 wct4xxp/*.ima
/usr/lib/hotplug/firmware; \
fi
if [ -d /lib/firmware ]; then \
/usr/bin/install -c -m 644 wct4xxp/*.ima /lib/firmware; \
fi
Installed firmware
/usr/bin/install -c -D -m 755 libtonezone.a /usr/lib/libtonezone.a
/usr/bin/install -c -D -m 755 libtonezone.so /usr/lib/libtonezone.so.1.0
if [ -z -a `id -u` = 0 ]; then \
/sbin/ldconfig || : ;\
fi
rm -f /usr/liblibtonezone.so
/bin/ln -sf libtonezone.so.1.0 \
/usr/lib/libtonezone.so.1
/bin/ln -sf libtonezone.so.1.0 \
/usr/lib/libtonezone.so
if [ -z ] [ -x /usr/sbin/sestatus ] (/usr/sbin/sestatus | grep
SELinux
status: | grep -q enabled) ; then restorecon -v /usr/lib/libtonezone.so;
fi
/usr/bin/install -c -D -m 644 zaptel.h /usr/include/zaptel/zaptel.h
/usr/bin/install -c -D -m 644 tonezone.h /usr/include/zaptel/tonezone.h
rm -f /usr/include/linux/zaptel.h
rm -f /usr/include/linux/torisa.h
rm -f /usr/include/zaptel.h
rm -f /usr/include/torisa.h
rm -f /usr/include/tonezone.h
if [ -f ztcfg ]; then \
/usr/bin/install -c -D -m 755 ztcfg /sbin/ztcfg; \
fi
if [ -f sethdlc-new ]; then \
/usr/bin/install -c -D -m 755 sethdlc-new /sbin/sethdlc; \
elif [ -f sethdlc ]; then \
/usr/bin/install -c -D -m 755 sethdlc /sbin/sethdlc; \
fi
if [ -f zttool ]; then \
/usr/bin/install -c -D -m 755 zttool /sbin/zttool; \
fi
for x in zaptel.ko ztd-loc.ko ztdummy.ko zttranscode.ko; do \
rm -f /lib/modules/2.6.19-1.2911.fc6xen/extra/$x ; \
done; \
make -C /lib/modules/2.6.19-1.2911.fc6xen/build
SUBDIRS=/usr/src/zaptel-
1.4.0 INSTALL_MOD_PATH= INSTALL_MOD_DIR=misc modules_install; \
if [ -f datamods/syncppp.ko ]; then \
make -C datamods install; \
else \
rm -f
/lib/modules/2.6.19-1.2911.fc6xen/misc/{hdlc_*,syncppp}.ko
; \
fi
make[1]: Entering directory `/usr/src/kernels/2.6.19-1.2911.fc6-x86_64'
INSTALL /usr/src/zaptel-1.4.0/zaptel.ko
INSTALL /usr/src/zaptel-1.4.0/ztd-loc.ko
INSTALL /usr/src/zaptel-1.4.0/ztdummy.ko
INSTALL /usr/src/zaptel-1.4.0/zttranscode.ko
DEPMOD 2.6.19-1.2911.fc6
So where wer ethey installed to?
Into /lib/modules/2.6.19-1.2911.fc6-x86_64 (no xen) by any chance?
In the worst case, copy them manually:
cp *.ko /lib/modules/`uname -r`/misc/
depmod -a
and hope for the best.
make[1]: Leaving directory `/usr/src/kernels/2.6.19-1.2911.fc6-x86_64'
if ! [ -f wcfxsusb.o ]; then \
rm -f /lib/modules/2.6.19-1.2911.fc6xen/misc/wcfxsusb.o; \
fi; \
rm -f /lib/modules/2.6.19-1.2911.fc6xen/misc/wcfxs.o
/usr/bin/install -c -m 644 doc/ztcfg.8 /usr/share/man/man8
/usr/bin/install -c -m 644 doc/zttool.8 /usr/share/man/man8
[ `id -u` = 0 ] /sbin/depmod -a 2.6.19-1.2911.fc6xen || :
[ -f /etc/zaptel.conf ] || /usr/bin/install -c -D -m 644 zaptel.conf.sample
/etc
/zaptel.conf
build_tools/genmodconf linux26 ztd-loc ztdummy zttranscode
Building /etc/modprobe.d/zaptel...
***
*** WARNING:
*** If you had custom settings in /etc/modprobe.d/zaptel,
*** they have been moved to /etc/modprobe.d/zaptel.bak.
***
*** In the future, do not edit /etc/modprobe.d/zaptel, but
*** instead put your changes in another file
*** in the same directory so that they will not
*** be overwritten by future Zaptel updates.
***
I cannot see any error.
The Warning is due to I already compiled zaptel several times, and
zaptel.conf, and zaptel already exists.
Carlos Alperin
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--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] analog channels calling out not detect DTMF
Jerry Geis wrote: I have a TDM2402E card. Occasionally I have noticed that a number I call that gives and IVR the DTMF keys are not detected. All other times the DTMF works fine. You'll probably want to increase your TX gains a little. I had the same issue until I did this. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk / ACT CRM Integration
Do You have a link to ACT CRM ? Thanks On 2/20/07, Cory Andrews [EMAIL PROTECTED] wrote: Has anyone ever been party to an integration of ACT CRM platform with Asterisk? Thanks Cory Andrews ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk / ACT CRM Integration
Hi Cory, I've never done it before but as I remember ACT has a TAPI interface so the connectivity should be pretty easy. Regards, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Tuesday, 20 February 2007 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk / ACT CRM Integration Has anyone ever been party to an integration of ACT CRM platform with Asterisk? Thanks Cory Andrews ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Inbound Problem
Well, could be the fact provider not pushing as g729 or someting else. Can you set debug 999 and set verbose 999 then redump that ? you are missing the before the answer part also.. Also try G711 first then work your way to other codecs On 2/20/07, Rajeev Natarajan [EMAIL PROTECTED] wrote: Am working with Arun on this project - here's a longer description of the problem: We've been fighting with our service provider on this issue - we seem to be getting a BYE just after we receive an ACK. They claim that it is an asterisk issue! The service provider provides only IP based authentication for inbound. We have used username-password based authentication with the same setup with *no problems* whatsoever! If we configure an Audiocodes MEdia gateway to receive the calls, there is no issue - so there's something that asterisk is doing? or asterisk-Provider gateway combo? In our efforts to mask IP, I have used PROVIDER-IP for the IP of my service provider (host) and AsteriskIP to indicate my asterisk server sip.conf [PROVIDER] type=peer disallow=all allow=g729 context=default host= fromuser=y.y.y.y port=5060 insecure=very canreinvite=no nat=yes qualify=yes CLI output: -- Executing Answer(SIP/PROVIDER-IP-b7a076a8, ) in new stack We're at 124.7.195.102 port 47698 Adding codec 0x100 (g729) to SDP Reliably Transmitting (NAT) to PROVIDER-IP:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bK6bd3121243ee9f936c4aeb96d6785b7a;received=PROVIDER-IP From: sip:[EMAIL PROTECTED];tag=3380976385-794612 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:8009422419@'AsteriskIP' Content-Type: application/sdp Content-Length: 183 v=0 o=root 2172 2172 IN IP4 AsteriskIP s=session c=IN IP4 AsteriskIP t=0 0 m=audio 47698 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - --- -- Executing Playback(SIP/PROVIDER-IP-b7a076a8, park) in new stack -- Playing 'park' (language 'en') AstSQL*CLI -- SIP read from PROVIDER-IP:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 5 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 From: sip:[EMAIL PROTECTED];tag=3380976385-794612 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Via: SIP/2.0/UDP 221.135.102.100:5060;branch=z9hG4bK02505a1dcc5937d9a648eebc0052b422 Content-Length: 0 --- (9 headers 0 lines) --- AstSQL*CLI -- SIP read from PROVIDER-IP:5060: BYE sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 5 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 From: sip:[EMAIL PROTECTED];tag=3380976385-794612 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE Via: SIP/2.0/UDP 221.135.102.100:5060 ;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f Content-Length: 0 --- (9 headers 0 lines) --- Sending to PROVIDER-IP : 5060 (NAT) Transmitting (NAT) to PROVIDER-IP:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f;received=PROVIDER-IP From: sip:[EMAIL PROTECTED];tag=3380976385-794612 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 The following is an ngrep of the traffic for an inbound call - 'U' marks the begin of the packet grabbed. U PROVIDER-IP:5060 - AsteriskIP:5060 INVITE sip:800942@AsteriskIP SIP/2.0..Max-Forwards: 5..Session-Expires: 3600;Refresher=uac..Suppor ted: timer..To: sip:[EMAIL PROTECTED]:5060..From: sip:PROVIDER-IP;tag=3380960452-790279..Co ntact: sip:PROVIDER-IP:5060..Remote-Party-Id: sip:PROVIDER-IP;party=calling;screen=no;privacy =off..Call-ID: [EMAIL PROTECTED]: 1 INVITE..Via: SIP/2.0/UDP 221. 135.102.100:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4..Allow-Events: telephone-event..Content-T ype: application/sdp..Content-Length: 206v=0..o=nextone-msw1 1774 4816 IN IP4 PROVIDER-IP..s=sip call..c=IN IP4 PROV-IP-2..t=0 0..m=audio 18932 RTP/AVP 18 19..a=ptime:20..a=rtpmap:19 CN/8000..a=fm tp:18 annexb=yes..a=rtpmap:18 G729/8000.. # U AsteriskIP:5060 - PROVIDER-IP:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4; received=PROVIDER-IP..From: sip:PROVIDER-IP;tag=3380960452-790279..To: sip:[EMAIL PROTECTED] 11.2:5060..Call-ID: [EMAIL PROTECTED]: 1 INVITE..User-Agent: Ast erisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:[EMAIL PROTECTED]..Content-Length: 0 # U AsteriskIP:5060 - PROVIDER-IP:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4
Re: [asterisk-users] Agents busy in queue
Yes first thing is not using 1.4 but as you probably won't budge , try hints. exten = 1001,hint,SIP/USER that will force it to poll status of that peer and reset the queue agent, of course replace values with actual ones On 2/20/07, Paul Hales [EMAIL PROTECTED] wrote: Are you using attended transfers? PaulH On Tue, 2007-02-20 at 15:37 +0500, Kashif Anwar wrote: I need some help with a problem which I'm facing with Asterisk 1.4 final release. I'm using static agents in a queue. Sometimes when an agent answers a call in queue and then releases it, the status for that agent in the queue remains busy where as there is not channel associated to that SIP client. For furthur calls in that queue that particular agent receives no more calls unless you unregister and then register that SIP client. This is occuring very regularly. Any one with a solution or idea?? Thanks, Kashif. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
With all other things said.. you might want a professional service for this like targusinfo.com Maintaining and running an operation like a cname web lookup thing is REALLY high overhead in terms of web traffic etc What happens when you get 30 ITSP/clients pulling 1000 calls each or 10 calls each per day.. that can easily go up to 1 mill requests per day , How will you pay for the bandwith/hardware/failover/load balance etc hardware for all this ? or if you are going to charge then why reinvent the wheel. targusinfo.com is what we would use.. Cname lookup is a really controversial matter , no one wants to absorb the costs , that is why some TELCOS charge 4.95 for callerid ( its basically the lookup service they are paying for) .. CNAME lookups is also not mandatory for TELCOS so some do it some don't , but FREE cname is just not going to happen untill some one has a Return on Investment strategy for this.. Take a look at Free 800 systems that went down , Any venture needs a capital source of income.. my 0.02 On 2/20/07, Robert Norton - SophMedia LLC [EMAIL PROTECTED] wrote: Hey Guys, I'm glad to see this ignited some discussion. I definitely understand there's some legal implications involved, both on a privacy level, and fraud prevention. Obviously an end-user (ie: the person controlling a listing) has to consent to some sort of release resolving the privacy concerns. I'm somewhat aware of the legal implications involved with storing such personally identifiable information (or whatever the legal term is) and have a concern in making sure such issues are resolved. In reality, how is it efficient for every provider to be running their own database? In my mind, this leaves the horribly evident inaccuracies, and even efficiency issues. Thank God these accuracies aren't integral to the operations of telephony systems. I do understand there is a price to pay for such infrastructure, and I believe that it's obvious the telephony world is riddled with racketeering, price gouging ventures, including companies that charge nearly a $0.01 for a lookup. I realize the following analogy is poor, but in mind this is as close as a internet search engine charging for a basic search query. Infact a basic internet query is much more complex, much more costly (ie: the infrastructure of said systems), and yet self-subsidizing. And to the poster who suggested that I was implying scrapping the results from 411.com, this is definitely not even a remote idea in my mind at all. The basis for my idea was a open, moderated, database that was user controlled and self-subsidized. I know this is way off topic, but I really feel that the telecom industry as a whole, and I'm sure I'm not the only one with this belief, is horribly bloated, running on business models that are clearly 30 years outdated. It is 2007, and with the help of the internet, the exchange of information, these telcos now have real, global competition, and real issues to deal with. Anyways guys, I'm curious to hear your thoughts. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
Why not make it like DNS and have each provider have their lookups deligated to a local server and then each ISP will run a caching server that will use a serial number system to get updates.. just like DNS. I know there are lot more DNS lookups then CNAM lookups per hour... isn't there? :) On 2/20/07, Mike Lynchfield [EMAIL PROTECTED] wrote: With all other things said.. you might want a professional service for this like targusinfo.com Maintaining and running an operation like a cname web lookup thing is REALLY high overhead in terms of web traffic etc What happens when you get 30 ITSP/clients pulling 1000 calls each or 10 calls each per day.. that can easily go up to 1 mill requests per day , How will you pay for the bandwith/hardware/failover/load balance etc hardware for all this ? or if you are going to charge then why reinvent the wheel. targusinfo.com is what we would use.. Cname lookup is a really controversial matter , no one wants to absorb the costs , that is why some TELCOS charge 4.95 for callerid ( its basically the lookup service they are paying for) .. CNAME lookups is also not mandatory for TELCOS so some do it some don't , but FREE cname is just not going to happen untill some one has a Return on Investment strategy for this.. Take a look at Free 800 systems that went down , Any venture needs a capital source of income.. my 0.02 On 2/20/07, Robert Norton - SophMedia LLC [EMAIL PROTECTED] wrote: Hey Guys, I'm glad to see this ignited some discussion. I definitely understand there's some legal implications involved, both on a privacy level, and fraud prevention. Obviously an end-user (ie: the person controlling a listing) has to consent to some sort of release resolving the privacy concerns. I'm somewhat aware of the legal implications involved with storing such personally identifiable information (or whatever the legal term is) and have a concern in making sure such issues are resolved. In reality, how is it efficient for every provider to be running their own database? In my mind, this leaves the horribly evident inaccuracies, and even efficiency issues. Thank God these accuracies aren't integral to the operations of telephony systems. I do understand there is a price to pay for such infrastructure, and I believe that it's obvious the telephony world is riddled with racketeering, price gouging ventures, including companies that charge nearly a $0.01 for a lookup. I realize the following analogy is poor, but in mind this is as close as a internet search engine charging for a basic search query. Infact a basic internet query is much more complex, much more costly (ie: the infrastructure of said systems), and yet self-subsidizing. And to the poster who suggested that I was implying scrapping the results from 411.com, this is definitely not even a remote idea in my mind at all. The basis for my idea was a open, moderated, database that was user controlled and self-subsidized. I know this is way off topic, but I really feel that the telecom industry as a whole, and I'm sure I'm not the only one with this belief, is horribly bloated, running on business models that are clearly 30 years outdated. It is 2007, and with the help of the internet, the exchange of information, these telcos now have real, global competition, and real issues to deal with. Anyways guys, I'm curious to hear your thoughts. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
TP'n to follow flow just like DNS, the 'root servers' would still see the high request hits, prior to passing off to local caching app. and *someone* must have this expense/headache to maintain them. Natambu Obleton wrote: Why not make it like DNS and have each provider have their lookups deligated to a local server and then each ISP will run a caching server that will use a serial number system to get updates.. just like DNS. I know there are lot more DNS lookups then CNAM lookups per hour... isn't there? :) On 2/20/07, Mike Lynchfield [EMAIL PROTECTED] wrote: With all other things said.. you might want a professional service for this like targusinfo.com Maintaining and running an operation like a cname web lookup thing is REALLY high overhead in terms of web traffic etc What happens when you get 30 ITSP/clients pulling 1000 calls each or 10 calls each per day.. that can easily go up to 1 mill requests per day , How will you pay for the bandwith/hardware/failover/load balance etc hardware for all this ? or if you are going to charge then why reinvent the wheel. targusinfo.com is what we would use.. Cname lookup is a really controversial matter , no one wants to absorb the costs , that is why some TELCOS charge 4.95 for callerid ( its basically the lookup service they are paying for) .. *snipped ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
TP'n to follow flow just like DNS, the 'root servers' would still see the high request hits, prior to passing off to local caching app. and *someone* must have this expense/headache to maintain them. No, the root servers wouldn't. Please take a few moments to learn how the domain name system works prior to spreading fear, uncertainty, and doubt. Any decent DNS server is extremely good at caching lookup responses, and as such, once it looks up the NS records for a domain, the roots and parents will not see a significant increase in requests. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Open CallerID Database?
ML == Mike Lynchfield [EMAIL PROTECTED] writes: ML With all other things said.. you might want a professional service ML for this like targusinfo.com ML Maintaining and running an operation like a cname web lookup thing ML is REALLY high overhead in terms of web traffic etc ML What happens when you get 30 ITSP/clients pulling 1000 calls each ML or 10 calls each per day.. You use DNS. ML that can easily go up to 1 mill requests per day , Not a problem with DNS. The technical problems are relatively easy to overcome. The other problems less so, probably. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Open CallerID Database?
RL == Richard Lyman [EMAIL PROTECTED] writes: RL TP'n to follow flow just like DNS, the 'root servers' would still RL see the high request hits, prior to passing off to local caching RL app. The DNS root servers are almost only loaded by irrelevant traffic. The root information is easily cacheable, so it is rare to have to actually ask the root servers. An ENUM-style solution would most likely not see much garbage traffic, and the relevant traffic is easily cacheable. I doubt that we will ever see such a solution though; there is too much invested in the old way of doing things. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Open CallerID Database?
Well caching is the way to go., bu then again most of the current solutions have this problem. John smit has a DID.. 514 555 1234 and closes account.. did sleeps for 3 months and new client Jane doe takes it.. Now how long should caching be ? this is a big problem ATM because some cache for 1 year others 1 day , they don't want to tell how long nor provider an API update method. On 20 Feb 2007 20:43:37 +0100, Benny Amorsen [EMAIL PROTECTED] wrote: RL == Richard Lyman [EMAIL PROTECTED] writes: RL TP'n to follow flow just like DNS, the 'root servers' would still RL see the high request hits, prior to passing off to local caching RL app. The DNS root servers are almost only loaded by irrelevant traffic. The root information is easily cacheable, so it is rare to have to actually ask the root servers. An ENUM-style solution would most likely not see much garbage traffic, and the relevant traffic is easily cacheable. I doubt that we will ever see such a solution though; there is too much invested in the old way of doing things. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
Tzafrir, You 'll going to love this one: There was no /lib/modules/2.6.19-1.2911.fc6xen/misc, and of course nothing as *.ko. But there was a /lib/modules/2.6.19-1.2911.fc6/misc with all the *.ko modules. I finished to copy the misc directory to the fc6xen, but still running modprobe zaptel I get: FATAL: Module zaptel not found. And on modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy The /lib/modules/2.6.19-1.2911.fc6/misc has all the .ko files, including a 'xpp' and a wct4xxp directory [EMAIL PROTECTED] misc]# ls pciradio.ko wcfxo.kowctdm24xxp.ko wcusb.ko ztd-eth.ko ztdynamic.ko tor2.ko wct1xxp.ko wctdm.ko xppztd-loc.ko zttranscode.ko torisa.kowct4xxp wcte11xp.kozaptel.ko ztdummy.ko Something get twisted on the directories, or the make generated the fc6/misc directory, and didn't take care of the Fc6xen directory, on /lib/modules. Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
Joe Greco wrote: TP'n to follow flow just like DNS, the 'root servers' would still see the high request hits, prior to passing off to local caching app. and *someone* must have this expense/headache to maintain them. No, the root servers wouldn't. Please take a few moments to learn how the domain name system works prior to spreading fear, uncertainty, and doubt. Any decent DNS server is extremely good at caching lookup responses, and as such, once it looks up the NS records for a domain, the roots and parents will not see a significant increase in requests. ... JG gee joe, what part of 'prior to passing off to local caching app', and millions of requests didn't you get? (also note, i understood they were wanting to build a new system, not use dns, as it was *only* the example) everytime you make a dns request, i agreed that it does not hit the root servers, but every time you request a NON-cached one you DO. so maybe your call center calls the same people every other day. ours do not, and i'm just guessing here, but i have to think that others here don't call the same people over and over and over millions of times within minutes/hours/days. yeah, you are right, i have no clue what i am talking about. don't you just hate when someone puts and apple in with the oranges, especially a rotten one. G ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk / ACT CRM Integration
I looked into it once. As far as I can tell they took out the TAPI interface a couple years ago. Probably too many support issues. Without a TAPI interface I would say it would not be very easy if at all possible/practical. -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 20, 2007 10:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk / ACT CRM Integration Hi Cory, I've never done it before but as I remember ACT has a TAPI interface so the connectivity should be pretty easy. Regards, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Tuesday, 20 February 2007 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk / ACT CRM Integration Has anyone ever been party to an integration of ACT CRM platform with Asterisk? Thanks Cory Andrews ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Open CallerID Database?
ML == Mike Lynchfield [EMAIL PROTECTED] writes: ML Well caching is the way to go., bu then again most of the current ML solutions have this problem. ML John smit has a DID.. 514 555 1234 and closes account.. did sleeps ML for 3 months and new client Jane doe takes it.. ML Now how long should caching be ? this is a big problem ATM because ML some cache for 1 year others 1 day , they don't want to tell how ML long nor provider an API update method. The actual records should have TTL's of a few hours, perhaps a day. The rest of the hierarchy can probably get away with longer TTL's, especially close to the root. It's the same thing with the root in regular DNS: You can cache the set of records in the root zone for a long long time, since noone is going to suddenly move all the .com nameservers. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rules about congestion
On my wild learning curve, I encountered numerous occasions when a channel remained in Congestion state after a Congestion() step without going to the next step, which is Hangup(). I couldn't find a definite pattern but it seems to happen when a channel is hung up by the other party or by some other action. Any recommendation about preventing such? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Open CallerID Database?
Am Dienstag, den 20.02.2007, 14:54 -0500 schrieb Mike Lynchfield: Well caching is the way to go., bu then again most of the current solutions have this problem. John smit has a DID.. 514 555 1234 and closes account.. did sleeps for 3 months and new client Jane doe takes it.. Now how long should caching be ? this is a big problem ATM because some cache for 1 year others 1 day , they don't want to tell how long nor provider an API update method. Coming back to the DNS example, there are certain timeouts. I have to admit I cannot tell how exactly the timeout values work together, but you _can_ set an absolute timeout after which any cached data (counted from the moment of retrieval) is marked obsolete and a subsequent query occurs. If you set something in the 2-week-range (which may or may not be what many people use in DNS) you can be pretty sure that freshly assigned numbers do not have dangling cache records, assuming the 3 months gap before assigning the same number again. Assuming one could add an additional TXT record to enum, say name.0.6.0.7.x.x.x.enum.info. TXT Hoffmeister, Anselm Martin this would pretty much do the trick. I have no idea wether any standard describes name resolution via enum. The other way around would be more tricky btw., with all those John Smith around ;) BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] analog channels calling out not detect DTMF
Doug, Thanks, right now my TX gain is 4.0 I thought I read somewhere not to go higher than 5. What are your thoughts? Jerry Jerry Geis wrote: / I have a TDM2402E card. // // Occasionally I have noticed that a number I call that gives and IVR // the DTMF keys are not detected. All other times the DTMF works fine. / You'll probably want to increase your TX gains a little. I had the same issue until I did this. Doug -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk / ACT CRM Integration
Could have, been years since I looked at ACT. I use salesforce.com exclusively for all my projects these days. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of shadowym Sent: Tuesday, 20 February 2007 3:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk / ACT CRM Integration I looked into it once. As far as I can tell they took out the TAPI interface a couple years ago. Probably too many support issues. Without a TAPI interface I would say it would not be very easy if at all possible/practical. -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 20, 2007 10:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk / ACT CRM Integration Hi Cory, I've never done it before but as I remember ACT has a TAPI interface so the connectivity should be pretty easy. Regards, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Tuesday, 20 February 2007 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk / ACT CRM Integration Has anyone ever been party to an integration of ACT CRM platform with Asterisk? Thanks Cory Andrews ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rules about congestion
Yuan LIU wrote: On my wild learning curve, I encountered numerous occasions when a channel remained in Congestion state after a Congestion() step without going to the next step, which is Hangup(). I couldn't find a definite pattern but it seems to happen when a channel is hung up by the other party or by some other action. Any recommendation about preventing such? Yuan Liu try using exten = xyz,x,SoftHangup(|a) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR reports short call length
Hello, In watching the console where two calls are natively bridged, Asterisk shows a hangup for each channel (both using IAX). This isn't correct and causes my CDR records to show shorter calls than what actually occurred. Before I go into higher detail, does anyone have any ideas about this? Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Tipping Point IPS blocking Asterisk SIP quaility messages
Hi guys, Just wanted to give you a heads up, so you don't end up chasing strange issues... Since early this morning, our Tipping Point IPS is blocking the Asterisk generated SIP Quality messages (the ones which tell you how good or badly reachably a remote SIP server is) Rule 5051: SIP: PROTOS Test Suite INVITE Test Case This filter detects a test case from the PROTOS SIP testing suite. PROTOS test suites are designed to fizz popular protocols to discover weaknesses in particular implementation. The PROTOS SIP test suite fuzzes SIP INVITE messages by sending several thousand combinations of illegal, abnormal, and overlong values for a variety of SIP INVITE message parameters. The results of these results range from unexpected responses to denial of service conditions to classic buffer overvlow error conditions. Vendor Site: http://.eee.oulu.fi/research/ouspq/protos/ It seems to be default to block, which will cause a couple of issues for people today :-) Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] analog channels calling out not detect DTMF
Jerry Geis wrote: Doug, Thanks, right now my TX gain is 4.0 I thought I read somewhere not to go higher than 5. What are your thoughts? Do you get complaints about low volume levels? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip to sip ?
Hi Rob.
The local * box works fine for all local sip calls to local sip calls
i have setup 2 voip handsets and they work well, even took one home
and tried it from my private nat'ed home network, all works, the
phones register and i can call the other extension, regardless of
location.
The * server is not firewalled at all and uses a public ip address.
the only problem seems to be that i can't call other * boxes or sip
users not local to my * box.
On 2/21/07, Rob Schall [EMAIL PROTECTED] wrote:
If you're getting a 404, I would assume it is reacting like any other
non-connection would (http, etc). Do you know if the packets are
reaching the phone, or if the phone is registering its correct IP
Address? If it is registering, but no packets are reaching it, could it
be a routing issue?
Rob
Chris Hills wrote:
Dennis Kavadas wrote:
hi all
i've just setup an * box and want to test voip calling, initially from
sip user to sip user...
local sip users can call each other, no issues.
problem arises when i try and call a remote sip account, my * box
always returns SIP/2.0 404 Not Found
any ideas ?
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Dennis
I use the following as my default context:-
[default]
exten = _X.,1,NoOp(Incoming Call from ${CALLERID} for
[EMAIL PROTECTED])
exten = _X.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten = _X.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten = _X.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
exten = _X.,5,GotoIf($[${SIPDOMAIN} = ${MYIP}]?10)
exten = _X.,6,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
exten = _X.,7,NoOp(@${SIPDOMAIN} is remote - forwarding...)
exten = _X.,8,Macro(uridial,[EMAIL PROTECTED])
exten = _X.,9,HangUp()
exten = _X.,10,Goto(default-noturi,${EXTEN},1)
exten = h,1,HangUp()
exten = s-BUSY,1,Congestion
exten = s-CHANUNAVAIL,1,Congestion
exten = s-CONGESTION,1,Congestion
[macro-uridial]
exten = s,1,NoOp(Outbound SIP URI call ${ARG1})
exten = s,2,SetCIDNum(0123456789)
exten = s,3,Dial(SIP/${ARG1})
exten = s,4,Congestion()
HTH
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Re: [asterisk-users] analog channels calling out not detect DTMF
You might try increasing the toneduration or whatever the option is in /etc/asterisk/zapata.conf Asterisk's default transmitted DTMF tone length is quite short. Jerry Geis wrote: Doug, Thanks, right now my TX gain is 4.0 I thought I read somewhere not to go higher than 5. What are your thoughts? Jerry Jerry Geis wrote: / I have a TDM2402E card. // // Occasionally I have noticed that a number I call that gives and IVR // the DTMF keys are not detected. All other times the DTMF works fine. / You'll probably want to increase your TX gains a little. I had the same issue until I did this. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't get ANSWEREDTIME after hangup using ZAP
Dear all, I tried to make a call with PHP AGI. $rc = execute_agi(EXEC DIAL ZAP/g1/$myphonenumber|60|rhHL( . ($max_total_seconds * 1000) . :6:3) ); $rc = execute_agi(GET VARIABLE ANSWEREDTIME ); And I can't get the answered time after caller hangup in this method. But if I use a SIP channel as below: $rc = execute_agi(EXEC DIAL SIP/$mysiptrunk/$myphonenumber|60|rhHL( . ($max_total_seconds * 1000) . :6:3) ); $rc = execute_agi(GET VARIABLE ANSWEREDTIME ); I can get the correct answered time. Is any idea about it? Is it the problem of my ZAP channel's configuration? -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Open CallerID Database?
RL == Richard Lyman [EMAIL PROTECTED] writes: RL everytime you make a dns request, i agreed that it does not hit RL the root servers, but every time you request a NON-cached one you RL DO. Nope. If you request foo.com and you have up to two days earlier visited bar.com, you won't hit the root servers. Only the .com servers. RL so maybe your call center calls the same people every other day. RL ours do not, and i'm just guessing here, but i have to think that RL others here don't call the same people over and over and over RL millions of times within minutes/hours/days. yeah, you are right, RL i have no clue what i am talking about. People have a tendency to call other people in the same area codes more often than people in other area codes. That ought to help load on the root servers. Anyway, a single server can easily handle 1000 queries per second. If you add even 0.1 cent to the call setup fee to pay for the lookup and you keep the servers at 100 qps average, you are looking at $8640 a day per server. Or look at it the other way around, if you allocate $1000 a month to run a server, and that server performs at 100 qps average, each call costs you .0004 cent extra. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Open CallerID Database?
Benny Amorsen wrote: RL == Richard Lyman [EMAIL PROTECTED] writes: RL everytime you make a dns request, i agreed that it does not hit RL the root servers, but every time you request a NON-cached one you RL DO. Nope. If you request foo.com and you have up to two days earlier visited bar.com, you won't hit the root servers. Only the .com servers. which would make it 'NON-cached' RL so maybe your call center calls the same people every other day. RL ours do not, and i'm just guessing here, but i have to think that RL others here don't call the same people over and over and over RL millions of times within minutes/hours/days. yeah, you are right, RL i have no clue what i am talking about. People have a tendency to call other people in the same area codes more often than people in other area codes. That ought to help load on the root servers. that is if your caller base are residential. call centers do not follow this. Anyway, a single server can easily handle 1000 queries per second. If you add even 0.1 cent to the call setup fee to pay for the lookup and you keep the servers at 100 qps average, you are looking at $8640 a day per server. Or look at it the other way around, if you allocate $1000 a month to run a server, and that server performs at 100 qps average, each call costs you .0004 cent extra. which if you want redundancy (like was mentioned already)... using the 'root servers' as a *model* someone would have to have this expense/headache to maintain /Benny i'm not sure why i seem to be unable to get my point across (even with multiple attempts), so i will just not try. good luck ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Passing a variable from one Asterisk box to another
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call dials into Box 2 the variable Foo does not get passed...
Does anyone have any clever ideas?
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RE: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
Tzafrir, After delete the /lib/modules/2.6.2.6.19-1.2911.fc6, previous to move the misc directory to /lib/modules/2.6.19-1.2911.fc6xen This is what I get on trying running modprobe zaptel ztdummy. [EMAIL PROTECTED] zaptel-1.4.0]# modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.19-1.2911.fc6xen/misc/zaptel.ko): Invalid module format [EMAIL PROTECTED] zaptel-1.4.0]# modprobe ztdummy WARNING: Error inserting zaptel (/lib/modules/2.6.19-1.2911.fc6xen/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.19-1.2911.fc6xen/misc/zaptel.ko): Invalid module format FATAL: Error inserting ztdummy (/lib/modules/2.6.19-1.2911.fc6xen/misc/ztdummy.ko): Invalid module format FATAL: Error running install command for ztdummy Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing a variable from one Asterisk box to another
Eric Bishop wrote:
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call dials into Box 2 the variable Foo does not get passed...
Does anyone have any clever ideas?
as noted in asterisk/docs/README.variables (iirc)
you should see that variable inheritance can occur by prefacing the
variable with '_' or '__'
also, depending on the age of your asterisk you might want to start
using 'Set' vice 'SetVar'
also, having ${EXTEN:0} , the :0 doesn't do anything, so you should not
use it and just have ${EXTEN}
i hope this helps
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Re: [asterisk-users] Passing a variable from one Asterisk box to another
Richard Lyman wrote:
Eric Bishop wrote:
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call dials into Box 2 the variable Foo does not get passed...
Does anyone have any clever ideas?
as noted in asterisk/docs/README.variables (iirc)
you should see that variable inheritance can occur by prefacing the
variable with '_' or '__'
also, depending on the age of your asterisk you might want to start
using 'Set' vice 'SetVar'
also, having ${EXTEN:0} , the :0 doesn't do anything, so you should
not use it and just have ${EXTEN}
i hope this helps
sadly replying to my own post, but, i forgot to mention that
passing variables with IAX2 can be an issue sometimes when you use
user and peer (the user side can pass vars the peer side can not, or
doesn't accept them iirc)
this does not happen using friend, but that has its own issues... check
the wiki for more thoughts about this.
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Re: [asterisk-users] Re: Open CallerID Database?
So how does this start? I mean it wouldn't be hard to modify dns server to use 3/3/4 format ip address... or it would need to be 3/3/3/4 for international right and someone wrote a module for asterisk look up that way and then I took my SS7 connection and setup a GTT gateway to a server so that real telco's could query it over SS7 and use my other CNAM provider as backup so that people could make me their main connection, but alas... most cnam providers don't allow you to resale lookups. Time to move on, but it would be kewl to start a revolution. :) On 2/20/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Dienstag, den 20.02.2007, 14:54 -0500 schrieb Mike Lynchfield: Well caching is the way to go., bu then again most of the current solutions have this problem. John smit has a DID.. 514 555 1234 and closes account.. did sleeps for 3 months and new client Jane doe takes it.. Now how long should caching be ? this is a big problem ATM because some cache for 1 year others 1 day , they don't want to tell how long nor provider an API update method. Coming back to the DNS example, there are certain timeouts. I have to admit I cannot tell how exactly the timeout values work together, but you _can_ set an absolute timeout after which any cached data (counted from the moment of retrieval) is marked obsolete and a subsequent query occurs. If you set something in the 2-week-range (which may or may not be what many people use in DNS) you can be pretty sure that freshly assigned numbers do not have dangling cache records, assuming the 3 months gap before assigning the same number again. Assuming one could add an additional TXT record to enum, say name.0.6.0.7.x.x.x.enum.info. TXT Hoffmeister, Anselm Martin this would pretty much do the trick. I have no idea wether any standard describes name resolution via enum. The other way around would be more tricky btw., with all those John Smith around ;) BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR reports short call length
Before I go into higher detail, does anyone have any ideas about this? Yes, see the transfer option for IAX. Set it to transfer=mediaonly which will leave the signaling unchanged and the channel alive, and thus produce correct CDRs. See: http://www.asterisk.org/doxygen/1.4/Config_iax.html PS: Never tried it... --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Passing a variable from one Asterisk box to another
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Bishop
Sent: Tuesday, February 20, 2007 5:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Passing a variable from one Asterisk box to
another
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro.
All works great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call dials into Box 2 the variable Foo does not get
passed...
Does anyone have any clever ideas?
The correct way using SIP is to add X headers before the Dial and then
pulling them in and assigning them to channel variables on the ingress
box. Here's a snippet that shows the idea:
On the box dialing out:
exten = _23XX,1,Set(Foo=1234) --- Use Set here not SetVar
exten = _23XX,2,SIPAddHeader(X-Foo: ${FOO})
exten = _23XX,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
On the ingress box:
exten = _23XX,1,Set(Foo=${SIP_HEADER(X-Foo)})
exten = _23XX,2,Answer()
...yada yada
Regards,
- Brad
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Re: [asterisk-users] Linux Command Line Soft Phone - $200+ bonus
--- Tzafrir Cohen [EMAIL PROTECTED] wrote: your requirement don't really make sense. to try again: speex and/or gsm - put into /etc/init.d/___ - phone enabled on boot up Huh? IS that phone a client program? If so: why should it be run as a server? There are plenty of ways to run a program at desktop startup. you are right - I'm after ease of maintenance. - automatically navigate around gnome and kde sound Huh? I have noticed in some soft phone docs that the gnome and kde sound systems need to be turned off for the soft phone to work. the phone needs to work with neither gnome nor kde running. - automatically navigate dhcp (if any) Huh? that the softphone will find its way to the server whether or not the computer is behind a router (dhcp). - gnu has something sort of close(?) - must install through one command, thru apt-get, or thru synaptic On which distribution? initial install on Ubuntu 6.10 Debian already has a host of free phones. The best seem to be Twinkle and Ekiga for SIP and kiax for IAX. A number of others are usable. The only one that does *both* SIP and IAX (if you really need that) is yate-gtk :-p . I'm after is a phone that is working when the computer boots. NO user interface - all control is done through the asterisk server to which it's connected. also, as above, not dependent on gnome or kde - with no graphics this should not be a problem. - must be hosted in free public place - must run on Ubuntu first try Which version? Ubuntu has some of the Debian packages. 6.10 phase 2: - have same run under Puppy Linux Consider giving more information on the limitations of the system (memory? disk-space?) I'm concerned (and ignorant) about installation Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Open CallerID Database?
Quoting Natambu Obleton [EMAIL PROTECTED]: So how does this start? I mean it wouldn't be hard to modify dns server to use 3/3/4 format ip address... or it would need to be 3/3/3/4 for international right and someone wrote a module for asterisk look up that way and then I took my SS7 connection and setup a GTT gateway to a server so that real telco's could query it over SS7 and use my other CNAM provider as backup so that people could make me their main connection, but alas... I think you have the format of the address space reversed conceptually but the idea would work - the countrycode, area code etc are like the .com in dns and then exchange like the 2ld, and number like the hostname. the only modification would be to set different root servers for this sort of parallel system, and have someone actually in charge to delegate the subdomains etc. registering a domain name has cost for the admin part of it, but is someone really going to pay to register a phone number they already pay for ? (the administration has to be paid for somehow) I am not trying to criticize, but just pointing out the realities of making it work. most cnam providers don't allow you to resale lookups. Time to move on, but it would be kewl to start a revolution. :) On 2/20/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Dienstag, den 20.02.2007, 14:54 -0500 schrieb Mike Lynchfield: Well caching is the way to go., bu then again most of the current solutions have this problem. John smit has a DID.. 514 555 1234 and closes account.. did sleeps for 3 months and new client Jane doe takes it.. Now how long should caching be ? this is a big problem ATM because some cache for 1 year others 1 day , they don't want to tell how long nor provider an API update method. Coming back to the DNS example, there are certain timeouts. I have to admit I cannot tell how exactly the timeout values work together, but you _can_ set an absolute timeout after which any cached data (counted from the moment of retrieval) is marked obsolete and a subsequent query occurs. If you set something in the 2-week-range (which may or may not be what many people use in DNS) you can be pretty sure that freshly assigned numbers do not have dangling cache records, assuming the 3 months gap before assigning the same number again. Assuming one could add an additional TXT record to enum, say name.0.6.0.7.x.x.x.enum.info. TXT Hoffmeister, Anselm Martin this would pretty much do the trick. I have no idea wether any standard describes name resolution via enum. The other way around would be more tricky btw., with all those John Smith around ;) BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Does Asterisk support DNIS?
Just to let everyone know. I restart my Asterisk box and ztcfg wouldn't run any more. I reran wancfg-zaptel and now everything is working correctly. It's picking up the DNIS digits without any problem. I still have to figure out ztcfg quits working every time I reboot though. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
I ran into this same problem compiling ndiswrapper on my MacBook with FC6. I uninstalled the old-time GCC and used yum to install gcc++3.3, I think it was and then re-compiled. FC6 has some wierdness with compiling. Ask me how fun it was getting everything working in my MacBook. -Original Message- From: Carlos Alperin [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 20, 2007 3:10 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64 Importance: High Tzafrir, After delete the /lib/modules/2.6.2.6.19-1.2911.fc6, previous to move the misc directory to /lib/modules/2.6.19-1.2911.fc6xen This is what I get on trying running modprobe zaptel ztdummy. [EMAIL PROTECTED] zaptel-1.4.0]# modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.19-1.2911.fc6xen/misc/zaptel.ko): Invalid module format [EMAIL PROTECTED] zaptel-1.4.0]# modprobe ztdummy WARNING: Error inserting zaptel (/lib/modules/2.6.19-1.2911.fc6xen/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.19-1.2911.fc6xen/misc/zaptel.ko): Invalid module format FATAL: Error inserting ztdummy (/lib/modules/2.6.19-1.2911.fc6xen/misc/ztdummy.ko): Invalid module format FATAL: Error running install command for ztdummy Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Open CallerID Database?
I would guess that registration would be by the telco for the blocks just like with reverse dns today, so then each telco would have a local server to manage their 'reverse' cnam lookup and the people in charge would be NANPA, just like how ARIN is regulated today. Although who owns the root namservers.. I wonder if ARIN and RIPE share ownership of them? Although now that i Think about it dns wouldn't work because want to deligate .. XXX-XXX-XYYY and XXX-XXX-XXYY and then there is single numbers. For that right now I do weird PTR CNAME to A record thing for single reverse dns. This would be little larger... ohh shit.. LNP. So now Qwest would need to deligate a single CNAM to me and crap..naw this will never work. On 2/20/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Natambu Obleton [EMAIL PROTECTED]: So how does this start? I mean it wouldn't be hard to modify dns server to use 3/3/4 format ip address... or it would need to be 3/3/3/4 for international right and someone wrote a module for asterisk look up that way and then I took my SS7 connection and setup a GTT gateway to a server so that real telco's could query it over SS7 and use my other CNAM provider as backup so that people could make me their main connection, but alas... I think you have the format of the address space reversed conceptually but the idea would work - the countrycode, area code etc are like the .com in dns and then exchange like the 2ld, and number like the hostname. the only modification would be to set different root servers for this sort of parallel system, and have someone actually in charge to delegate the subdomains etc. registering a domain name has cost for the admin part of it, but is someone really going to pay to register a phone number they already pay for ? (the administration has to be paid for somehow) I am not trying to criticize, but just pointing out the realities of making it work. most cnam providers don't allow you to resale lookups. Time to move on, but it would be kewl to start a revolution. :) On 2/20/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Dienstag, den 20.02.2007, 14:54 -0500 schrieb Mike Lynchfield: Well caching is the way to go., bu then again most of the current solutions have this problem. John smit has a DID.. 514 555 1234 and closes account.. did sleeps for 3 months and new client Jane doe takes it.. Now how long should caching be ? this is a big problem ATM because some cache for 1 year others 1 day , they don't want to tell how long nor provider an API update method. Coming back to the DNS example, there are certain timeouts. I have to admit I cannot tell how exactly the timeout values work together, but you _can_ set an absolute timeout after which any cached data (counted from the moment of retrieval) is marked obsolete and a subsequent query occurs. If you set something in the 2-week-range (which may or may not be what many people use in DNS) you can be pretty sure that freshly assigned numbers do not have dangling cache records, assuming the 3 months gap before assigning the same number again. Assuming one could add an additional TXT record to enum, say name.0.6.0.7.x.x.x.enum.info. TXT Hoffmeister, Anselm Martin this would pretty much do the trick. I have no idea wether any standard describes name resolution via enum. The other way around would be more tricky btw., with all those John Smith around ;) BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] analog channels calling out not detect DTMF
eric, This is what I am trying to find. How to make those durations longer. I didnt really find anything in indications although it seems like it should be there. Does anyone know how to increase the DTMF tone duration? jerry You might try increasing the toneduration or whatever the option is in /etc/asterisk/zapata.conf Asterisk's default transmitted DTMF tone length is quite short. Jerry Geis wrote: / Doug, // // Thanks, right now my TX gain is 4.0 // I thought I read somewhere not to go higher than 5. // // What are your thoughts? // // Jerry // // Jerry Geis wrote: // / I have a TDM2402E card. // // // // Occasionally I have noticed that a number I call that gives and IVR // // the DTMF keys are not detected. All other times the DTMF works fine. // / // You'll probably want to increase your TX gains a little. I had the same // issue until I did this. // // Doug // // / Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] analog channels calling out not detect DTMF
Read the zapata.conf.sample file that comes with Asterisk [EMAIL PROTECTED] ~]# grep toneduration /etc/asterisk/zapata.conf toneduration=300 ;toneduration=100 [EMAIL PROTECTED] ~]# Jerry Geis wrote: eric, This is what I am trying to find. How to make those durations longer. I didnt really find anything in indications although it seems like it should be there. Does anyone know how to increase the DTMF tone duration? jerry You might try increasing the toneduration or whatever the option is in /etc/asterisk/zapata.conf Asterisk's default transmitted DTMF tone length is quite short. Jerry Geis wrote: / Doug, // // Thanks, right now my TX gain is 4.0 // I thought I read somewhere not to go higher than 5. // // What are your thoughts? // // Jerry // // Jerry Geis wrote: // / I have a TDM2402E card. // // // // Occasionally I have noticed that a number I call that gives and IVR // // the DTMF keys are not detected. All other times the DTMF works fine. // / // You'll probably want to increase your TX gains a little. I had the same // issue until I did this. // // Doug // // / Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] analog channels calling out not detect DTMF
TP'n to follow flow or mod the /etc/asterisk/indications.conf the /xxx is the duration (iirc) example: busy is like 400/400,0/400 the /400 (each) is the duration Eric ManxPower Wieling wrote: Read the zapata.conf.sample file that comes with Asterisk [EMAIL PROTECTED] ~]# grep toneduration /etc/asterisk/zapata.conf toneduration=300 ;toneduration=100 [EMAIL PROTECTED] ~]# Jerry Geis wrote: eric, This is what I am trying to find. How to make those durations longer. I didnt really find anything in indications although it seems like it should be there. Does anyone know how to increase the DTMF tone duration? *snipped ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] analog channels calling out not detect DTMF
/etc/asterisk/indications.conf has nothing to do with the length of DTMF tones sent out FXO ports. Richard Lyman wrote: TP'n to follow flow or mod the /etc/asterisk/indications.conf the /xxx is the duration (iirc) example: busy is like 400/400,0/400 the /400 (each) is the duration Eric ManxPower Wieling wrote: Read the zapata.conf.sample file that comes with Asterisk [EMAIL PROTECTED] ~]# grep toneduration /etc/asterisk/zapata.conf toneduration=300 ;toneduration=100 [EMAIL PROTECTED] ~]# Jerry Geis wrote: eric, This is what I am trying to find. How to make those durations longer. I didnt really find anything in indications although it seems like it should be there. Does anyone know how to increase the DTMF tone duration? *snipped ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] analog channels calling out not detect DTMF
sorry, i read 'detect DTMF' which i took over 'calling out'... so, i thought since he already made a reference to 'not finding anything in indications', i would offer this tidbit. if nothing else it will help the *next person* that does a ML search for 'duration detect dtmf'. Eric ManxPower Wieling wrote: /etc/asterisk/indications.conf has nothing to do with the length of DTMF tones sent out FXO ports. Richard Lyman wrote: TP'n to follow flow or mod the /etc/asterisk/indications.conf the /xxx is the duration (iirc) example: busy is like 400/400,0/400 the /400 (each) is the duration Eric ManxPower Wieling wrote: Read the zapata.conf.sample file that comes with Asterisk [EMAIL PROTECTED] ~]# grep toneduration /etc/asterisk/zapata.conf toneduration=300 ;toneduration=100 [EMAIL PROTECTED] ~]# Jerry Geis wrote: eric, This is what I am trying to find. How to make those durations longer. I didnt really find anything in indications although it seems like it should be there. Does anyone know how to increase the DTMF tone duration? *snipped ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] analog channels calling out not detect DTMF
eric THanks, I was grep'ing for toneduration. tone, duration and nothing. It wasnt in my file for some reason. ANyway, Thanks, I'll give it a try. Jerry Read the zapata.conf.sample file that comes with Asterisk [root at pbx-1 http://lists.digium.com/mailman/listinfo/asterisk-users ~]# grep toneduration /etc/asterisk/zapata.conf toneduration=300 ;toneduration=100 [root at pbx-1 http://lists.digium.com/mailman/listinfo/asterisk-users ~]# Jerry Geis wrote: / eric, // // This is what I am trying to find. How to make those durations longer. // I didnt really find anything in indications although it seems like it // should be there. // // Does anyone know how to increase the DTMF tone duration? // // jerry // // // You might try increasing the toneduration or whatever the option is in // /etc/asterisk/zapata.conf Asterisk's default transmitted DTMF tone // length is quite short. // // Jerry Geis wrote: // / Doug, // // // Thanks, right now my TX gain is 4.0 // // I thought I read somewhere not to go higher than 5. // // // What are your thoughts? // // // Jerry // // // Jerry Geis wrote: // // / I have a TDM2402E card. // // // // // // Occasionally I have noticed that a number I call that gives and IVR // // // the DTMF keys are not detected. All other times the DTMF works // fine. // // / // // You'll probably want to increase your TX gains a little. I had the // same // issue until I did this. // // // Doug // // // / // Jerry // / ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] analog channels calling out not detect DTMF
We found that if you set it to +20, you get a lot of distortion. ;) (yes, we did do this once to see what would happen) PaulH On Tue, 2007-02-20 at 15:31 -0500, Jerry Geis wrote: Doug, Thanks, right now my TX gain is 4.0 I thought I read somewhere not to go higher than 5. What are your thoughts? Jerry Jerry Geis wrote: / I have a TDM2402E card. // // Occasionally I have noticed that a number I call that gives and IVR // the DTMF keys are not detected. All other times the DTMF works fine. / You'll probably want to increase your TX gains a little. I had the same issue until I did this. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Open CallerID Database?
Am Dienstag, den 20.02.2007, 16:33 -0700 schrieb Natambu Obleton: I would guess that registration would be by the telco for the blocks just like with reverse dns today, so then each telco would have a local server to manage their 'reverse' cnam lookup and the people in charge would be NANPA, just like how ARIN is regulated today. Although who owns the root namservers.. I wonder if ARIN and RIPE share ownership of them? Although now that i Think about it dns wouldn't work because want to deligate .. XXX-XXX-XYYY and XXX-XXX-XXYY and then there is single numbers. For that right now I do weird PTR CNAME to A record thing for single reverse dns. This would be little larger... ohh shit.. LNP. So now Qwest would need to deligate a single CNAM to me and crap..naw this will never work. I do not get your point here. Take ENUM. A made-up phone number like +49 (228) 91234567 would be found as 7.6.5.4.3.2.1.9.8.2.2.9.4.enum-something This means, you delegate the 2.1.9.8.2.2.9.4.enum-something to the telco that owns the 912 block in Bonn, Germany. Number portability screws this, as even single numbers out of a contiguous range of MSNs on a ISDN line can be moved over to another provider, with the others staying with the old provider. They would have to play well together, and that will fubar for sure (they even sometimes block the DSL frequencies on lines when a customer moves to another company, with unblocking taking a 14-day security period, because they can - there is no technical block, but the DSLAM manipulation database software will mark your DSL line as blocked, making moving to another provider a pain with up to 6 weeks without internet). In Germany we have an agency that manages all phone book data, and (nearly?) all the 411 type services (called 118xx here) buy data there. How they manage their data internally is none of my business (although I would guess it's something like MSSQL with an Access frontend, thinking about the Deutsche Telekom ;-). They will no way accept DNS-type queries for free. Some months ago, there was a heise.de (German IT newsticker) article about them charging enourmous sums, paying the real data storage and administration costs back by about factor 5 or so. Well-paying businesses rarely give away their business turnpoint. Any non-official system will suffer even worse inaccuracy than the providers' own and managed system (as someone else already wrote). Their data is quite bad enough. This relates, of course, to the fact that they may only reverse-lookup numbers to find names if the customer explicitely allowed them to do it, on the line rental agreement. There are usually several checkboxes, allowing you to get listed in phonebook, get listed in digital listings, and get listed for reverse lookup. For those who allow it there is a free web-frontend to reverse-lookup numbers, which is a pain to script-access, but it is possible. It suffers from problems with DIDs, as for example a shop might have the number 94144-0, and assigned the numbers up to 94144-29. If you try to lookup 9414488 (which might be a private person's analogue line, and this is absolutely valid in the German numbering system) it will return the wrong entry because the logic in their webinterface always assumes that 94144-0 means all numbers starting 94144- belong to that line. _That_ really sucks - you think business and then it's a friend calling for private talk. Just out of interest: From former posts I understood that there is a CALLERID service in US (for an extra fee, I assume) that gives both number _and_ name of the caller...? I am aware of the fact that e.g. EuroISDN lines can transmit alphanumeric callerid (and in fact I already use that on an ISDN phone here that connects to an Asterisk - showing a few special names like wakeup call). Not for names yet, as I was too lazy to implement that. Does that also work over analogue lines? BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] They ignore my DTMF!
Hello, I can call out to the PSTN and talk to people but when I have to enter a dtmf tone in an ivr or voicemail system those systems do not recognise that I have sent a tone. This is the case when I make the call with the Xlite softfone or a regular telephone plugged into a PAP2NA or a Linksys SPA941. However... a Grandstream GXP2000 works just great ??? All are extensions on my Asterisk 1.4 box. I am using a voip trunk through Atlasvoice. All extensions are setup identical in sip.conf. One last thing, if a system wants me to respond 1 for sales 2 for service I can hit the 1 button quickly 4 or 5 times and the remote system will get it. That does not work for a three digit extension as you may well imagine. Any help would be appreciated. Pierre ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rules about congestion
On Tue, 2007-02-20 at 11:05 -0800, Yuan LIU wrote: On my wild learning curve, I encountered numerous occasions when a channel remained in Congestion state after a Congestion() step without going to the next step, which is Hangup(). I couldn't find a definite pattern but it seems to happen when a channel is hung up by the other party or by some other action. Any recommendation about preventing such? If you look at the following page: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Congestion You can see there the following: Sends a signal to inform the channel of congestion. This command waits for the user to hang up; it does not continue execution of further commands. new in asterisk 1.2: Now this app supports an optional 'timeout' argument. If the optional timeout is specified, the calling channel will be hung up after the specified number of seconds. Otherwise, this application will wait until the calling channel hangs up. A little google or a search on the above site will answer most of your questions. -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rules about congestion
From: Carlos Chavez [EMAIL PROTECTED] Date: Tue, 20 Feb 2007 19:12:42 -0600 On Tue, 2007-02-20 at 11:05 -0800, Yuan LIU wrote: On my wild learning curve, I encountered numerous occasions when a channel remained in Congestion state after a Congestion() step without going to the next step, which is Hangup(). I couldn't find a definite pattern but it seems to happen when a channel is hung up by the other party or by some other action. Any recommendation about preventing such? If you look at the following page: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Congestion You can see there the following: Sends a signal to inform the channel of congestion. This command waits for the user to hang up; it does not continue execution of further commands. Thanks, Carlos, for the reference. Guess I was misled by the many sample codes that have Congestion() followed by Hangup() to believe that I need to signal congestion before hanging up. I'm starting to take Congestion() out when I want to Hangup() immediately. But congestion condition can occasionally go beyond timeout. If it is a SIP channel, I'll have a hard time even soft hang it because show channels cuts channel name short. Yuan Liu new in asterisk 1.2: Now this app supports an optional 'timeout' argument. If the optional timeout is specified, the calling channel will be hung up after the specified number of seconds. Otherwise, this application will wait until the calling channel hangs up. A little google or a search on the above site will answer most of your questions. -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chávez Prats Director de TecnologÃa +52-55-91169161 ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk CDR MySQL
I'm attempting to setup Asterisk 1.4.0 CDRs to use MySQL. Modules show like cdr_mysql.so tells me it is loaded. Reload cdr with MySQL started or stopped makes no difference in the errors. Ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
