[Asterisk-Users] spandsp-0.0.2pre11
Hi, People often send me audio logs from spandsp's soft-fax machine, where they have problems with corruption in the middle of a page for most or all of their faxes. Their problems are usually due to frame slips. However, recently I have received audio logs from two people who generally have great success with my software, but who find one or two machine regularly fail to send faxes correctly to their Asterisk boxes. I have looked at these logs and found the timing difference between tne near and far modems is out of spec. Well, out of spec or not, it must be tolerated. I have modified my V.29 modem to tolerate larger timing errors, and made this version available as spandsp-0.0.2pre11. Please try it. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 x dual g729 channels x other choices?
I confirm too, Sipura devices have flawless g729a codec. Tested personally the Sipura-2100, 3000 and 841 hardphone models - all work with Asterisk 100% straight out of the box, even with chan_sip's not_so_100%_rfc3261 behaviour. I think the sipura-1001 model is the stripped-down 1 fxs port copy of 2100, and as they all share the same firmware core, it should work ok as well. The overheating problem seems gone too, my sipuras are only moderately warm after running continously for several days. Now if Sipura could make an 8 port fxs version... ;-) Michael D Schelin wrote: Don't believe everything you read. There is nothing wrong with the sound quality of the G729 codec on the sipura devices. The 2000 does not support both channels running G729 at the same time. This limitation has be fixed with there new product. I forget the model number. Most G729 sound problems can be traced to busy or poorly designed networks. Too much packet loss. I'm a sip service provider and have seen everything with sip. Supura is the best product on the market today. Hermann Wecke wrote: begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: * - SMS w/out PSTN
Jay, Some Sender-ID issues came up during our beta trials but AFAIK these have all been fixed now in the live version. Also regarding MWI, keep an eye on your email in the next few days... Cheers... -- Adam Holt Bayham Systems Ltd Web:http://www.bayhamsystems.com/asterisk.html Email:adam.holt at bayhamsystems.co.uk Address: No. 1 Farnham Road, Guildford, Surrey, GU2 4RG, United Kingdom *** NEWS: Bayham Systems delivers first ever Global SMS Auction for UNICEF / GSM-Association. See how this raised over EUR90,000 at: http://www.gsm.org/ -Original Message- From: Jay Milk [mailto:jay at skimmilk.net] Sent: Thursday, March 24, 2005 16:04:09 CST 2005 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * - SMS w/out PSTN Worked pretty well last time I tried it, but they had issues submitting the sender-id, and they were unable to format a MWI-message. Prices aren't all that competitive, you'll have to purchase 100,000 credits before, or 33,333 SMSs before you hit the 10c/message barrier. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP images
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Chris W wrote: In a sense this cound be off-topic but I hope it isn't considered so. Apologies already if it is! Can anyone point me in the right direction to get new SIP images for the Cisco 7960 phone? I found P0S30202 around (ie v2.02) and it works but lacks a lot of the features the phone boasts so I'm looking for updates. I googled and found that you can get a support contract via 1-800-INSIGHT but guess what! They're in the US and won't issue licences outside the country. I'm in the Netherlands so that ain't gonna make matters easy. I guess I need v.3, 4, 5, 6 and 7 to get the latest stuff. What a lot of upgrading! Any pointers/help most welcome. Thanks in advance Unfortunatley, all the Cisco resellers in Europe I have approached don't seem to be interested in carrying these low value contracts (CON-SNT-CP7960 or CON-SNT-ATA186) or don't want to deal in such low volumes and have no method of dealing with such sales. Cisco want you to talk to their resellers, which brings you back right where you started. So to summarise: 1/ Cisco will not sell direct. 2/ North American Resellers will not sell to Europe. 3/ European Resellers do/will not sell single contracts What route is left for guy with a few Cisco phones in Europe? Piracy? /RANT - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD:519961 Gossiptel:9309811 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBQkfGYEtP/KMNOfRbAQKAZwgAi1QXt7d9Igy4o2dHG+qqG6KApixH01Xu x2lns+WvPwuDcHF5uBzJjfxGG4jVrgLtIg1la7M6P8Bu6u2nZQyz0fJk8UhVN4bp drsXHmjq44UyDel9Kn2Q6zvhfuND84qZTBAQ9MbLXnogQlg9vB067975P8rQ7+vK WX598aP0i5tDDMvhUNVZX/epYuIby0E6YdLwGaARcpcERWiQfG2tkY9EcVots1qt rcruHJZO4yutOwIY6irzmMpCShj+SShfRwNiI4+ggJIchUnaq+Il4ly4nMbDl2Px 5EZBWzECnQPRxeatKKyngZXUbMcFm9FozgLP7eHMol73QwlbWDjqfQ== =toHG -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pass caller ID to another application or machine.
On 18:57, Sun 27 Mar 05, Richard Reina wrote: I would like to have asterisk pass along the caller ID phone number to a database server on a my local network (the same network that the * server resides on ) so that our customer service app. can pull up customer data automatially. Asterisk passes along caller ID to the phones fine, can someone tell me how to make it pass this info to my database server? Hi, You can do this with an agi script. We are doing this in our app too. We have a webbased crm app and * looks up the number there and inserts a record into a table so our app can read that. When the call hangs up, the record is deleted from the database. It's not really that hard to make. -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't Dial Out with TDM04B
I am a beginer trying to install my first TDM04B.
Hi RR,
[outgoing]
exten = _0X.,1,Zap/1/${EXTEN}
I cant send them out.
The error is telling you that ZAP is not an application. To dial out
you need the dial application exactly as you have in the incoming
section. Something like this:
exten = _0X,1,Dial(ZAP/1/${EXTEN},45) ; for example
Please take a look at http://asteriskdocs.org
or look on the wiki for information about the asterisk dialplan:
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
Understanding the dialplan is critical to using asterisk.
hth
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[Asterisk-Users] TDM04B doesn't hang up after Voicemail
Hello all,
I am having a serious problem installing my * with a TDM04B. I made everything
work, call are coming in and going out including using a GSM Box in channel
Zap/2-1. I did setup voicemail like this on extensions.conf:
[incoming]
exten = s,1,Dial(SIP/2246,20)
exten = s,2,Wait,2
exten = s,3,Voicemail(u${ME})
exten = s,4,Hangup
exten = s,102,Wait,2
exten = s,103,Voicemail(b${ME})
exten = s,104,Hangup
After the call is finished if the user doesn't press # the line hangs forever.
Unfortunately I found it out after i did a zap show... 26 minutes after
the call ended :(. I looked into the threads but no answer seems to resolve
the problem (maxthreashhold or maxsilence and there is even a patch to one of
the voicemail files which i have no idea how to implement). The other strange
thing it is happening is that after i hang up the call from the phone if the
outside caller hasn't hang up it recreates the Zap channel and rings it
again.any clues please?
Thanks for the help.
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Re: [Asterisk-Users] TDM04B doesn't hang up after Voicemail
Have a look at http://www.voip-info.org/wiki-Asterisk+Disconnect+Supervision Julian J. M. On Mon, 28 Mar 2005 11:21:09 +, Robson Ribeiro [EMAIL PROTECTED] wrote: After the call is finished if the user doesn't press # the line hangs forever. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on a dialup connection?
Kerry Garrison wrote: Firefly supports Speex too but trying it just now I am getting no audio. -Kerry Type show translation in your Asterisk console to check if you have speex installed. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bug fixes IPSwitchBoard
Hi all, I have just released IPSwitchBoard version 0.70. There are no major changes, but a few important bug fixes. You can download IPS from the new website I have created for IPSwitchBoard: http://ipswitchboard.thorben.dk Regards Thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to park/transfer a call received from a Queue?
From: Matias G. [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] How to park/transfer a call received from a Queue? To: Asterisk Users Mailing List - Non-Commercial Discussion you haven't include hte part where you make AgentCallBackLogin() the context you enter there is the one where your call will be tried to place when the agent transfers it ie: exten = 11,1,AgentCallbackLogin(|[EMAIL PROTECTED]) will log that agent in a valid extension inside that context. when the agent tries to transfer he will be allowd to transfer to extensions valid in that context... hope this helps. GREAT! This was the trick! I just needed to add include = parkedcalls In the context of [CallCenter] include = parkedcalls Exten = .. All the phone extensions. And now it's parking and transfering as a charm :-)) Thanks Matias and the other hints I received from the list :-) Wessel de Roode -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.4 - Release Date: 27-03-05 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem parsing unusual SIP/SDP
The next step would to be turn pedantic=yes back on, then generate a failing call with 'sip debug', 'set verbose 255' and 'set debug 255' in place. Capture all the output (there will be a lot) and then post a bug in Mantis describing the situation and attaching the output file. Kevin, thanks again for the help. I now understand why it's not working, but don't know enough to suggest a fix, or even to say what routine has the bug. The problem relates to the additional checking done by find_call when pedantic=yes. In response to the original INVITE, the provider sends a challenge with a tag: SIP/2.0 401 UnAuthorized [other headers] f:Test User sip:[my phone [EMAIL PROTECTED];tag=as5822c02a t:sip:[dest [EMAIL PROTECTED];tag=1628255942721615 WWW-Authenticate: Digest ... [other headers] Asterisk saves the tag in the theirtag member of the sip_pvt structure and issues a new INVITE with suitable credentials. The provider initiates the call and returns progress: SIP/2.0 183 Session Progress [other headers] f:Test User sip:[my phone [EMAIL PROTECTED];tag=as5822c02a t:sip:[dest [EMAIL PROTECTED];tag=e5559e9a-1dd1-11b2-b48e-b03162323164+e5559e9a Well, provider is now sending a different tag, so Asterisk does not find a match, assumes that this response is for a call it does not know about, and discards it. Although this is ugly SIP, one can understand why it would happen, and IMHO it is legal. RFC 3261 says: When the originating UAC receives the 401 (Unauthorized), it SHOULD, if it is able, re-originate the request with the proper credentials. I believe that re-originate means that we are starting a new dialog and the old tag should be discarded. However, I don't know where or when this should be done. In fact, I don't understand why the tag checking happens on outgoing calls at all. A comment in chan_sip.c says: /* In principle Call-ID's uniquely identify a call, however some vendors (i.e. Pingtel) send multiple calls with the same Call-ID and different tags in order to simplify billing. The RFC does state that we have to compare tags in addition to the call-id, but this generate substantially more overhead which is totally unnecessary for the vast majority of sane SIP implementations, and thus Asterisk does not enable this behavior by default. Short version: You'll need this option to support conferencing on the pingtel */ That makes sense, but since Asterisk always generates a unique Call-ID for each call, I would think that tag checking on outgoing calls would be unnecessary. However, the routine carefully chooses the From or To field according to the call direction, so there is probably a good reason to check all calls. Indeed, the change that I would request might break operation with some other provider or device. Is it worth posting such a vague bug report? Unfortunately, I know absolutely nothing about the internals of Asterisk. Thanks, Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323: g711-g729 transcoding
I have a connect to * via H.323/g711 from device A and want to connect to B which want for H.323/g729 h323.conf contains disallow=all allow=alaw allow=g729 but outgoing faststart/TCS contains only g711 (from h323_request(format) i think) and so no codec negotiation and no voice. Howto run up g711/H323 - * - g729/H323 PS intel's g729 was used. ast 1.0.3-6 PPS stupid - h323_set_capability(format/*=8*/, dtmfmode); + h323_set_capability(capability/*=8+256 (711a+729)*/, dtmfmode); lead to segv only. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323: g711-g729 transcoding
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Orehov Pasha wrote: I have a connect to * via H.323/g711 from device A and want to connect to B which want for H.323/g729 h323.conf contains disallow=all allow=alaw allow=g729 but outgoing faststart/TCS contains only g711 (from h323_request(format) i think) and so no codec negotiation and no voice. Howto run up g711/H323 - * - g729/H323 You probably also need to allow ulaw Change h323.conf to: disallow=all allow=alaw allow=ulaw allow=g729 - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD:519961 Gossiptel:9309811 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBQkfr7ktP/KMNOfRbAQJP/QgAmsoI/vXDly/gc/ET8IJ1sOrewyad3UXJ jdyx+CS04rmc9egSc11ZZwZKGH+pJh1gGlnTo5k0Jib5yJx5Jrbsj7a3ccppM6GD dN10mV78Cp0zVydnJj/PBVK4ysXVrEE0DqysUZMRXLP7Kd54cC8w7S1/QXSxZGIE cgNIYM3eekiMzrCwXcyrz5u+oiajRn7Eqqolj5Q/HkN4/SxneIfUu+jH9aL1tDbT 4UZrhtqntPEfyAVnUxlKmgj6A0FyIxNUnxCwpkWD99f4VxgHkIYGJWuNhKX7YWg3 qWAGp8DGeIVD0nuyaKNUjgK8qZhj1DzMMR0+/pYRwp8PAf78WyeATA== =wk8m -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pass caller ID to another application or machine.
You can do this with an agi script. We are doing this in our app too. We have a webbased crm app and * looks up the number there and inserts a record into a table so our app can read that. When the call hangs up, the record is deleted from the database. It's not really that hard to make. DO you happen to rember the name of the agi command that thansfers the record into the table? Or do you know where I can find some sample sripts to look at? Thanks, RIchard __ Do you Yahoo!? Yahoo! Sports - Sign up for Fantasy Baseball. http://baseball.fantasysports.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP images
On Monday 28 March 2005 09:54, Ron Wellsted wrote: [...] So to summarise: 1/ Cisco will not sell direct. 2/ North American Resellers will not sell to Europe. 3/ European Resellers do/will not sell single contracts What route is left for guy with a few Cisco phones in Europe? Piracy? /RANT I don't think http://www.s2s.ltd.uk/ care how little you buy. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04B doesn't hang up after Voicemail
I have the same problem in my setup here, the TDM card deosn't detect
line hangup or any call progress related frequencies as they are
different for every country (does anybody know the progress frequencies
in Egypt?).
So my workaround for this was to add maxsilence and maxmessage in the
voicemail.conf, and add a hangup after the voicemail line in
extensions.conf, thus voicemail will detect silence and if it doesn't it
doesn't leave the line picked up for an infinite period of time.
Ezabi
Robson Ribeiro wrote:
Hello all,
I am having a serious problem installing my * with a TDM04B. I made everything
work, call are coming in and going out including using a GSM Box in channel
Zap/2-1. I did setup voicemail like this on extensions.conf:
[incoming]
exten = s,1,Dial(SIP/2246,20)
exten = s,2,Wait,2
exten = s,3,Voicemail(u${ME})
exten = s,4,Hangup
exten = s,102,Wait,2
exten = s,103,Voicemail(b${ME})
exten = s,104,Hangup
After the call is finished if the user doesn't press # the line hangs forever.
Unfortunately I found it out after i did a zap show... 26 minutes after
the call ended :(. I looked into the threads but no answer seems to resolve
the problem (maxthreashhold or maxsilence and there is even a patch to one of
the voicemail files which i have no idea how to implement). The other strange
thing it is happening is that after i hang up the call from the phone if the
outside caller hasn't hang up it recreates the Zap channel and rings it
again.any clues please?
Thanks for the help.
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Re: [Asterisk-Users] TDM04B doesn't hang up after Voicemail
maxsilence in voicemail.conf
- Original Message -
From: Robson Ribeiro [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, March 28, 2005 6:21 AM
Subject: [Asterisk-Users] TDM04B doesn't hang up after Voicemail
Hello all,
I am having a serious problem installing my * with a TDM04B. I made
everything
work, call are coming in and going out including using a GSM Box in
channel
Zap/2-1. I did setup voicemail like this on extensions.conf:
[incoming]
exten = s,1,Dial(SIP/2246,20)
exten = s,2,Wait,2
exten = s,3,Voicemail(u${ME})
exten = s,4,Hangup
exten = s,102,Wait,2
exten = s,103,Voicemail(b${ME})
exten = s,104,Hangup
After the call is finished if the user doesn't press # the line hangs
forever.
Unfortunately I found it out after i did a zap show... 26 minutes after
the call ended :(. I looked into the threads but no answer seems to
resolve
the problem (maxthreashhold or maxsilence and there is even a patch to one
of
the voicemail files which i have no idea how to implement). The other
strange
thing it is happening is that after i hang up the call from the phone if
the
outside caller hasn't hang up it recreates the Zap channel and rings it
again.any clues please?
Thanks for the help.
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Re: [Asterisk-Users] TDM01B
Does someone have a working config file they could send me? In /etc/zaptel.conf put something like this: defaultzone=us fxsks=1 loadzone=us where =1 is the fxo module for the pstn line. (I don't recall for sure, but if the fxo module is in module position #4, then I think you'll need fxsks=4 in the above.) If you are running cvs-head, then modprobe wctdm ztcfg -vvv You can run 'zttool' to see the card/module at this point. Once you have success (as indicated by the output of ztcfg), then: in /etc/asterisk/zapata.conf you'll need an entry something like: context=inbound-bus signalling=fxs_ks echocancel=yes echotraining=800 echocancelwhenbridged=no rxgain=0.0 txgain=0.0 immediate=no callprogress=no channel = 1 where channel = 1 must match the =1 number used in the /etc/zaptel.conf file. If you make any changes to /etc/asterisk/zapata.conf, you will need to stop asterisk and restart it; don't use the CLI reload. If you get to this point, then incoming calls on that fxo port will be sent to the inbound-bus context in your /etc/asterisk/extensions.conf file. An entry in that file something like: [inbound-bus] exten = s,1,Dial(SIP/3000,15) exten = s,2,Hangup will cause the sip phone at extension 3000 to ring for 15 seconds. If the zaptel/wctdm drivers are loaded, then 'cat /proc/interrupts' should show something like this: CPU0 0: 430422405 XT-PIC timer 1:389 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 0 XT-PIC usb-uhci 7:8801048 XT-PIC wctdm 8: 1 XT-PIC rtc 9: 18589687 XT-PIC ehci-hcd, eth0 where wctdm is on an interrupt by itself. If it is not, then move the TDM card to another slot on the motherboard to get it onto an unshared interrupt. If you run 'lsmod', you should see something like: Module Size Used byNot tainted wcusb 20096 0 (unused) wctdm 38144 4 zaptel179168 12 [wcusb wctdm] indicating both zaptel and wctdm are loaded, and zaptel is Used By the wctdm module. I don't use the stable version of asterisk. I recall someone posting something about the wctdm driver is actually wcfxo in stable. So, if the 'modprobe wctdm' complains, try 'modprobe wcfxo'. If you're in a non-US country, you may need to add the 'opermode=' parameter to config the driver for your country pstn standards. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pass caller ID to another application or machine.
On 04:07, Mon 28 Mar 05, Richard Reina wrote:
DO you happen to rember the name of the agi command
that thansfers the record into the table? Or do you
know where I can find some sample sripts to look at?
Hi,
Here are some snippets from my php-based agi:
#!/usr/bin/php4 -q
?php
ob_implicit_flush(true);
set_time_limit(6);
$in = fopen(php://stdin,r);
$stdlog = fopen(/var/log/asterisk/my_agi.log, w);
// toggle debugging output (more verbose)
$debug = false;
// Do function definitions before we start the main loop
function read() {
global $in, $debug, $stdlog;
$input = str_replace(\n, , fgets($in, 4096));
if ($debug) fputs($stdlog, read: $input\n);
return $input;
}
// parse agi headers into array
while ($env=read()) {
$s = split(: ,$env);
$agi[str_replace(agi_,,$s[0])] = trim($s[1]);
if (($env == ) || ($env == \n)) {
break;
}
}
.snip
// lookup id and name
$sql = SELECT a.id,a.bedrijfsnaam FROM adres as a,bcards as b WHERE
a.id=b.bedrijfs_id AND (replace(replace(a.telnr,'-',''), ' ','') ILIKE
'%.$agi[callerid].');
.snip
// put the stuff into our table
$sql = INSERT INTO active_calls VALUES
('.$row[bedrijfsnaam].',.$row[id].,.mktime().);
.snip
?
have fun!
--
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Two of the most famous products of Berkeley are LSD and BSD. I don't think
that this is a coincidence.
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RE: [Asterisk-Users] Realtime mysql problem?
++---+-+--+---+--+-- ---+---+- Here goes it's going to be messy :-) I followed the directions off the wiki. This *should* work just fine right? I built the table according to the directions, every field is varchar though, could that be a problem? Thanks again, Matt +---+--+--++-+--+--- --+-+-++-+-+ ++---++---+--+-- --++---++ | id | name | accountcode | amaflags | callgroup | callerid | canreinvite | context | peercontext | defaultip | dtmfmode | fromuser | fromdomain | host| insecure | mailbox | nat | pickupgroup | port | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret | type | username | disallow | allow | regseconds | ipaddr | cancallforward | ++---+-+--+---+--+-- ---+---+-+---+--+--+ +-+--+-+-+-+ +-+-+++---+- ---+---+--+++--- ++ | 3 | brak-test | [NULL] | [NULL] | [NULL]| [NULL] | no | outbound | incoming| [NULL]| [NULL] | [NULL] | [NULL] | dynamic | [NULL] | [NULL] | 0 | [NULL] | 4569 | [NULL] | [NULL] | [NULL] | [NULL] | blah | friend | brak-test | all | ulaw;alaw;g729 | 1112015162 | 206.80.254.254| yes | ++---+-+--+---+--+-- ---+---+-+---+--+--+ +-+--+-+-+-+ +-+-+++---+- ---+---+--+++--- ++ Matt Schulte wrote: Flatfile meaning iax.conf? Yes.. Sounds like a data problem to me. Paste your iaxpeers/iaxusers table schema and iax.conf section that is relevant to the phone. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pass caller ID to another application or machine.
Michiel,
Thanks very much for the resonse. I am confused
however by fopen(/var/log/asterisk/my_agi.log
my * system has not such log file only the Master.cvs
which only seems to log a call one its teminated?
--- Michiel van Baak [EMAIL PROTECTED] wrote:
On 04:07, Mon 28 Mar 05, Richard Reina wrote:
DO you happen to rember the name of the agi
command
that thansfers the record into the table? Or do
you
know where I can find some sample sripts to look
at?
Hi,
Here are some snippets from my php-based agi:
#!/usr/bin/php4 -q
?php
ob_implicit_flush(true);
set_time_limit(6);
$in = fopen(php://stdin,r);
$stdlog = fopen(/var/log/asterisk/my_agi.log,
w);
// toggle debugging output (more verbose)
$debug = false;
// Do function definitions before we start the main
loop
function read() {
global $in, $debug, $stdlog;
$input = str_replace(\n, , fgets($in, 4096));
if ($debug) fputs($stdlog, read: $input\n);
return $input;
}
// parse agi headers into array
while ($env=read()) {
$s = split(: ,$env);
$agi[str_replace(agi_,,$s[0])] = trim($s[1]);
if (($env == ) || ($env == \n)) {
break;
}
}
.snip
// lookup id and name
$sql = SELECT a.id,a.bedrijfsnaam FROM adres as
a,bcards as b WHERE a.id=b.bedrijfs_id AND
(replace(replace(a.telnr,'-',''), ' ','') ILIKE
'%.$agi[callerid].');
.snip
// put the stuff into our table
$sql = INSERT INTO active_calls VALUES
('.$row[bedrijfsnaam].',.$row[id].,.mktime().);
.snip
?
have fun!
--
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key:
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Two of the most famous products of Berkeley are LSD
and BSD. I don't think that this is a coincidence.
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RE: [Asterisk-Users] ADIT 600 Dynamic Impedance matching
Thanks for the response, it's a rather simple setup. What worries me is we're going into an old PBX, the channelbank goes 25pair about 20 feet to a punchdown block. Then from the block goes to another block (standard telco room layout) then to the phone system. The old phone system is a Meridian, about 20 years old. All the phones coming off that are analog from what I gather, the building wiring can range from 5 - 50 years old. What's unusual is I've never heard this echo personally. I've had the customer call from different phones of course and I've dialed out from these phones to even my cell phone and haven't had a problem. What's odd is this seems to be random, if I could get it to happen everytime on a single phone then I could point fingers at the internal wiring. shrug, else all I have to blame is the cb or the wiring between it and the pbx. Thoughts? -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 23, 2005 8:31 AM On March 23, 2005 08:25 am, Matt Schulte wrote: Has anyone ever heard of this so called Dynamic Impedance matching on the ADIT 600? I called their support and they've never heard of it. We That's odd, I have always had excellent support from CAC. And FWIW I've never had echo problems with their channel banks. Ever. I have echocancel turned off in the Zapata driver. The only clue to the dynamic impedance is that the 5g and ver8 of the FXS cards can hardcode the impedance according to country. Well that's fine and dandy but so can a Rhino CB-24 in the rating of milliamps.. You don't tune impedance in milliAmps. That's a current measurement. The Rhino can probably alter the amount of current it can source and this is what they're talking about. Not having used Rhino's stuff, I can't say for certain, but you simply don't alter impedance by changing mA. (yes, IAAEE). Does anyone have suggestions regarding these issues? Please hold back the flaming comments. I'm not here to flame, but to resolve and very tiring issue. :-) You can start by giving us a connection diagram between the Adit600 and whatever you're hooked up to, including grade of cable, how long it is, what it's terminating to (make and model) and whether you've tried replacing some runs with other cable to test. Invariably my Adit600 analogue runs are always under 50 feet since I'm terminating to a PBX or KSU nearby. These devices are able to terminate very long (km) runs, so I am curious as to why you're having such issues. Do you have the gains on the Adit600 or Zapata turned way up? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small qos switch
On Sunday 27 March 2005 22:30, Jim Sturtevant wrote: What product from Sangom and at what price point? Thx See original poster below. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of steve szmidt Sent: Sunday, March 27, 2005 6:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] small qos switch On Sunday 27 March 2005 13:48, Jim Sturtevant wrote: How about considering the linksys WRT54G (approx $59) with SVEASOFT firmware ($29) www.sveasoft.com which provides QOS by port, IP, and/or traffic type plus VPN, SNMP, etc. and WiFi to boot. Maybe because the Sangoma card will run circles around it purely from a performance view, never mind the quality. Which is usually important to a business... You can buy 400 series servers from Dell for around $350, new. Run your firewall (iptables) and NAT on that computer. You can get a Sangoma DSL PCI card for about $115--it has QoS. You'll have professional grade infrastructure for not that much money. What's not elegant about that? -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] another voipjet question
Haven't done this yet Art but I will try it today at the
office...Thanks Jonathan
On Mon, 28 Mar 2005 00:30:32 -0600, Tim Litwiller [EMAIL PROTECTED] wrote:
so where did you put these lines?
exten = _1NXXNXX,1,SetCallerID(4153574000)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011.,1,SetCallerID(4153574000)
exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
I want asterisk to use my pots line for local calls and voipjet for long
distance after the initial 100 free minutes my line provider give with
our plan. but to failover if one is busy and the other isn't.
Art Zemon wrote:
Jon Walsh wrote:
No Dice so far, anyone now how to add anIAX trunk? What are the
settings exactly?
Jon,
It took me awhile to get voipjet working with AAH because I was stubborn
and wanted to get it going through the AMP interface, instead of by hand
crafting the .conf files.
The trick was that I had to make *two* trunks for voipjet. The second
trick was to ignore a buglette in AMP. Here is what I did:
1. Create an IAX trunk. You *must* enter trunk name [EMAIL PROTECTED]
where 1234 is your voipjet ID. Cut 'n' paste all of the other
details from voipjet's site into the outgoing peer details window.
Leave all of the incoming stuff and the registration string blank;
you can't receive calls through voipjet.
2. Create a second IAX trunk. You *must* name this trunk voipjet. I
entered all of the same info here, too, but I think that all you
need in the peer details is the host= line.
If you don't create the second trunk, you will get a message in the log
that is something like voipjet: host not found when * tries to dial
with the string IAX2/[EMAIL PROTECTED]/16365551212
The buglette is that if you try to re-edit the first trunk, the first
digit from the trunk name will be missing. Fear not, the trunk name is
stored correctly THE FIRST TIME YOU SAVE. After that, AMP will mess it
up and you will need to remember to manually correct the trunk name if
you edit and save.
Cheers,
-- Art Z.
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[Asterisk-Users] BroadVoice - Failed to authenticate on INVITE error
I'm experiencing a Failed to authenticate on INVITE error (see output below) whenever I try to MAKE a call through the Broadvoice account. I noticed some others had the same problem but it went away when they rebuilt Asteris w/ a new version. N such luck for me! I'd be grateful for any assitance. Here's what I've done so far: 1) I downloaded the latest stable version of Asterisk and compiled it (27-Mar-05). 2) I updated my conf files as per the Broadvoice web site (see below) 3) I CAN make and recive calls through the Broadvoice account using X-Lite. 4) To avoid typos, I used cut and paste in sip.conf to copy the phone number and password from the register line to the [sip.broadvoice.com] section 5) When I run Asterisk: i) The Broadvoice account registers OK ii) I can receive calls on the Broadvoice account iii) I CANNOT make calls through the Broadvoice account. When I do, my computer freezes up but eventually comes around a while after I hangup and warns - Failed to authenticate on INVITE to 'asterisk sip:[EMAIL PROTECTED];tag=as4a325b3a' (see below) Any ideas? Finally, I'm still unclear about assigning an extension to the Broadvoice account as part of the registration line (see where I commented out ;/3003). What does it do? I rely on the context defined under the [sip.broadvoice.com] section. What do you gain by assigning an extension in the Register line? My conf files and the Asterisk output are below. Thanks, Jewel ;* ;/etc/hosts # Do not remove the following line, or various programs # that require network functionality will fail. 127.0.0.1 localhost.localdomain localhost # proxy.dca.broadvoice.com 147.135.0.128 sip.broadvoice.com ; ;* ; ;/etc/asterisk/sip.conf ; [general] port=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) context=from-sip-external ; Send unknown SIP callers to this context pedantic=no register = [EMAIL PROTECTED]:password:[EMAIL PROTECTED];/3003 ; [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=8145551212 secret=password username=8145551212 insecure=very context=from-broadvoice authname=8145551212 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no ; ;* ; /etc/asterisk/extensions.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here. ; ; [from-broadvoice] exten = s,1,Dial(ZAP/1,30) exten = s,2,Hangup [from_FXS] exten = _1NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30) exten = _1NXXNXX, 2, congestion() exten = _1NXXNXX, 102, busy() ; ;* ; ;/etc/asterisk/zapata.conf ; ; This is the bare bones of what is required to get your X100P ; card working on a normal line provided by a local phone ; carrier in North America. For more details on all options, ; see /usr/src/asterisk/configs/zapata.conf.sample but I would ; strongly suggest starting simple with the bare minimum of ; configs and working up from there - PSTN telephony interfaces ; are notoriously touchy with the large number of features ; they offer. ; [channels] language=en context=from-FXO signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 4 ; language=en context=from_FXS signalling=fxo_ks channel=1 ; language=en context=from-ILS-FXS signalling=fxo_ksFailed to authenticate on INVITE to 'asterisk sip:[EMAIL PROTECTED];tag=as4a325b3a' channel=2 ; ;* ; ;/Asterisk Console Output ; Asterisk Ready. *CLI sip show registry Host Username Refresh State 147.135.0.128:50608145551212 120 Registered *CLI -- Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] Mar 27 20:55:26 NOTICE[1116941248]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to 'asterisk sip:[EMAIL PROTECTED];tag=as4a325b3a' Mar 27 20:55:26 WARNING[1209214528]: app_dial.c:347 wait_for_answer: Unable to forward voice == Spawn extension (from_FXS, 13035551212, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why even use SIP
On Sat, 26 Mar 2005 04:14:54 +0200, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Mar 23, 2005 at 04:37:02PM -0500, Dana Olson wrote: My company has thousands of entries in the DHCP server, and it would take forever to go through each and every one of them. Not to mention that I, being in the telecom division, do not have access to the DHCP servers. scan for a MAC address? ping all the addresses in the range and then /usr/sbin/arp -n |grep -i that_mac_addr The scanning part could be done using something like: nmap -sP 192.168.1-5.* Another simple trick (assuming a mostly windows network) is to simply ping to the broadcast address. Linux-es and macs tend to respond to those pings and so are most devices. Windows tend to ignore those pings. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend The MAC addresses are not labeled on the units. I swear I said that already. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small qos switch
Although I'm not that familiar with it, I have heard good things about... http://www.bsdmall.com/saadpcico.html Don't know about hardware QOS on it tho... I'm assuming just shaping via the host machine. J On Mon, 28 Mar 2005 08:26:38 -0500, steve szmidt [EMAIL PROTECTED] wrote: On Sunday 27 March 2005 22:30, Jim Sturtevant wrote: What product from Sangom and at what price point? Thx See original poster below. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of steve szmidt Sent: Sunday, March 27, 2005 6:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] small qos switch On Sunday 27 March 2005 13:48, Jim Sturtevant wrote: How about considering the linksys WRT54G (approx $59) with SVEASOFT firmware ($29) www.sveasoft.com which provides QOS by port, IP, and/or traffic type plus VPN, SNMP, etc. and WiFi to boot. Maybe because the Sangoma card will run circles around it purely from a performance view, never mind the quality. Which is usually important to a business... You can buy 400 series servers from Dell for around $350, new. Run your firewall (iptables) and NAT on that computer. You can get a Sangoma DSL PCI card for about $115--it has QoS. You'll have professional grade infrastructure for not that much money. What's not elegant about that? -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pass caller ID to another application or machine.
On 05:14, Mon 28 Mar 05, Richard Reina wrote: Michiel, Thanks very much for the resonse. I am confused however by fopen(/var/log/asterisk/my_agi.log my * system has not such log file only the Master.cvs which only seems to log a call one its teminated? Richard, I created that file myself. That way I can put debug information into that logfile while developing that agi script. It's part of my skeleton agi script ;) You can safely remove it if you want. -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP images
Ron Wellsted wrote: What route is left for guy with a few Cisco phones in Europe? Piracy? I looked around for nearly a year for a contract after a kind soul got me the images (the closest I got was a site in the US who were prepared to sell me the CON-SNT-CP7960 for £8 ... with £150 Postage!!!)... eventually gave up and ordered a CON-SNT-PKG1 package from lanway which I managed to get for £42. Of course being a Cisco contract it still hasn't arrived 2.5 weeks later. Cisco are the first company I've ever come across who seem to actively resent having customers and would rather you went with someone else. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADIT 600 Dynamic Impedance matching
On March 28, 2005 08:15 am, Matt Schulte wrote: Thanks for the response, it's a rather simple setup. What worries me is we're going into an old PBX, the channelbank goes 25pair about 20 feet to a punchdown block. Then from the block goes to another block (standard telco room layout) then to the phone system. The old phone system is a Meridian, about 20 years old. All the phones coming off that are analog from what I gather, the building wiring can range from 5 - 50 years old. Yeah that's all pretty standard. What's unusual is I've never heard this echo personally. I've had the customer call from different phones of course and I've dialed out from these phones to even my cell phone and haven't had a problem. What's odd is this seems to be random, if I could get it to happen everytime on a single phone then I could point fingers at the internal wiring. shrug, else all I have to blame is the cb or the wiring between it and the pbx. Until you are able to recreate it it's going to be hard to nail down... I'd start by testing individual lines -- is it always line 3 that echoes? If the Meridian's hunting you may get the same line 5 times in a row or you may get it only when the moon is in Saturn's realm... And is it only specific destination numbers or ...? There are still too many variables. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange problems IAX / Monitor / ChanSpy CVS HEAD
From: Matias G. [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, March 27, 2005 11:21 PM Subject: [Asterisk-Users] Strange problems IAX / Monitor / ChanSpy CVS HEAD Hi list, I'm having some strange problems since I updated to CVS HEAD three hours ago... First: I was using Iax Comm in some PCs, it suddenly stopped working, what I get is som pieces of audio once in a while, I mean instead of listening to the ring tone and then the voice on the other side I just hear a bit of the ring tone, maybe another bit, a bit of someone answering... like ..beep(beep )...be(ep)...he(llo)...(hello)...(do you) hear m(e) (of course what's inside () is what I don't hear but can pretty well imagine) If I monitor these calls what I get is a -out.wav file with a normal size and a HUGE (really big like 2 Gb in 30 seconds) -in.wav file and last but not least when I try to ChanSpy a SIP channel (at last that's what I moved to the last CVS Head for) it makes * crash see what goes on (I was attached to an asterisk running using asterisk -vvvr) Mar 27 23:12:41 WARNING[5631]: app_chanspy.c:280 start_spying: Attaching SIP/matt2-05b0 to SIP/matt-967b Godzilla*CLI Disconnected from Asterisk server Executing last minute cleanups Godzilla:/etc/asterisk# I'm using Asterisk CVS-HEAD-03/27/05-17:22:56 currently running on Godzilla (pid = 5700) thanks a lot. M. sorry upon my insistence on this but I'm about to post a bug in mantis but would like to know if someone else is having the same problem... thanks again, M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI STREAM FILE command
Has anyone had success with the AGI STREAM FILE command with the CVS? I can't get it to work with the debian 1.0.5 package or the CVS on Redhat or Debian. It's not syntax, I'm doing that right. It doesn't give me an error when I use AGI DEBUG, it doesn't even give a response, just goes right on to the next command. I put a SAY NUMBER 123 # before and after the STREAM FILE and they both work fine, returning 200 OK, etc. [EMAIL PROTECTED] wrote: Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: How to use multiple VOIP provider trunks (Damon Estep) 2. RE: Asterisk on a dialup connection? (Kerry Garrison) 3. Re: How to use multiple VOIP provider trunks (Tim Pushor) 4. Re: Comedian Voicemail Issues (Matias G.) 5. RE: How to use multiple VOIP provider trunks (Damon Estep) 6. How to park/transfer a call received from a Queue? (Wessel de Roode) 7. pass caller ID to another application or machine. (Richard Reina) 8. RE: How to park/transfer a call received from aQueue? (Damon Estep) 9. Re: How to use multiple VOIP provider trunks (Tim Pushor) 10. Re: Asterisk on a dialup connection? (Tim Pushor) 11. RE: pass caller ID to another application or machine. (Damon Estep) 12. RE: How to use multiple VOIP provider trunks (Damon Estep) 13. Re: How to park/transfer a call received from aQueue? (Matias G.) 14. Re: pass caller ID to another application or machine. (C F) 15. RE: Asterisk on a dialup connection? (Kerry Garrison) 16. RE: small qos switch (Jim Sturtevant) 17. Re: TDM01B (Russell Handorf) 18. Re: Sipura 2000 x dual g729 channels x other choices? (Daniel Bruce Lynes) 19. Re: Sipura 2000 x dual g729 channels x other choices? ([EMAIL PROTECTED]) 20. Re: Sipura 2000 x dual g729 channels x other choices? (Andres) 21. Re: Sipura 2000 x dual g729 channels x other choices? (Andres) 22. Broadvoice getting unregistered (Courtney Couch) 23. RE: Broadvoice getting unregistered (Kerry Garrison) 24. Re: Asterisk on a dialup connection? (Saul Diaz) 25. Re: High Availability on Asterisk (Matthew Boehm) 26. Re: Broadvoice getting unregistered (Courtney Couch) 27. another voipjet question (Tim Litwiller) 28. Re: another voipjet question (Art Zemon) 29. Re: High Availability on Asterisk (Andres) -- Message: 1 Date: Sun, 27 Mar 2005 19:45:38 -0700 From: Damon Estep [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How to use multiple VOIP provider trunks To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii snip I am working on a phone routing system (with duplicate/redundant routes) and I will just have a way for a user to tell the system that they want to use an alternate route for the next call. How about the simple and traditional method, Dial 9 for an outside line, dial 8 for an alternate outside line? Or dial nothing for an outside line, dial 9 for an alternative outside line. -- Message: 2 Date: Sun, 27 Mar 2005 18:46:02 -0800 From: Kerry Garrison [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk on a dialup connection? To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Dialup quality is going to be very very poor to the point of not being usable most of the time. You should use a service that has a low bandwidth codec that works well like Skype or Teleo. The Codecs for Asterisk do not like dialup. I have heard that Speex might work ok but I havent tried it. Only Firefly supports it as far as I know. Kerry http://geekgazette.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Sunday, March 27, 2005 6:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk on a dialup connection? How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required unless I can use the onbord G723.1 codecs on the
[Asterisk-Users] Connecting quadbri to EuroISDN with 2 TE and 2 NT ports - what cables and settings ?
Hi, I'm trying to connect quadbri between powered ISDN phone and ISDN line: ISDN ---1--- TE - * - NT --2-- Phone I use quadbri, suse 9.2 and latest 0.2.0-RC7k bristuff. I've used sample settings provided with package, but do get strange error (I think that I have wrong setting for P2P or P2MP setting and cables 1 and 2). If I connect phone to ISDN with straight cable it works. I've put quadbri in between, and connected ISDN to span1 in TE mode, and phone in NT mode on span4. Did configuration (added at the end). I get errors: qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 I'm not sure what cables to use. I use straight for -1- and -2-. Is this right ? I'm in European community (EuroISDN) so I guess I should have some pretty standard connections... I have ISDN line with 2 MSNs and two connectors on NT termination - so I guess this is P2MP. Is this OK ? Does anyone have working example for Germany ? There is also one strange thing: I get this in dmesg when loading qozap module, although I have specified 4th port to be NT - and pri show span 4 shows it in netowork mode: Zapata Telephony Interface Unloaded module zaptel unsupported by SUSE/Novell, tainting kernel. Zapata Telephony Interface Registered on major 196 module qozap unsupported by SUSE/Novell, tainting kernel. PCI: Enabling device :02:0c.0 ( - 0003) ACPI: PCI interrupt :02:0c.0[A] - GSI 20 (level, low) - IRQ 209 qozap: S/T ports: 4 [ TE TE TE TE ] qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Registered tone zone 3 (Netherlands) Thanks in advance, regards, Rob. #- /etc/zaptel.conf: loadzone=nl defaultzone=nl # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 #- /etc/asterisk/zapata.conf [channels] switchtype = euroisdn pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres=yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 ;--- ; p2p TE mode (for connecting ISDN lines in point-to-point mode) ;signalling = bri_cpe ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp context=isdn-incoming group = 1 ; S/T port 1-3 (first quadBRI, or lower ports of an octoBRI) channel = 1-2 channel = 4-5 channel = 7-8 ;--- ; p2p NT mode (for connecting an ISDN PBX in point-to-point mode) signalling = bri_net context=pbx-incoming group = 2 ; S/T port 4 (second quadBRI, or upper ports of an octoBRI) channel = 10-11 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gsm player for Linux?
Hello, Does anyone know of an audio player for Linux that will play asterisk's GSM files? I know I can convert them using sox, but I'm hoping to play them natively so I can test out some voicemail settings. Thanks! -- Jesse Guardiani, Systems Administrator WingNET Internet Services, P.O. Box 2605 // Cleveland, TN 37320-2605 423-559-LINK (v) 423-559-5145 (f) http://www.wingnet.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can a sip.conf stanza be shared by several phones?
Hi, If several phones register to the same sip.conf section what will happen with a Dial SIP/shared in asterisk? All phones ringing and the first one to answer gets the call? Undefined behavior? Thanks, -- Jesus is coming! Everyone look busy! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime mysql problem?
Matt Schulte wrote: ++---+-+--+---+--+-- ---+---+- Here goes it's going to be messy :-) I followed the directions off the wiki. This *should* work just fine right? I built the table according to the directions, every field is varchar though, could that be a problem? The value of nat should be no or yes, not 0 (zero). Try that and reload everything. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI/Dialing out
Hi,
after having read so much about Asterisk, I went on and tried out to
create a little sample-setup.
I'm using a Fritz Card USB with the AVM Capi Driver and two X-Lite
Softphones.
Dialing between the softphones makes no problem.
Calling the MSN fron an external phone also works. I'm getting to the
asterisk demo-voicebox which works flawlessly.
Now may next step has been to enable dialing out with the softphones.
This does not work as expected.
I can dial out and the hard phone on the other end actually rings. When
I answer it, I can hear nothing. Noting appears on the Asterisk console,
X-Lite still talks about trying to connect.
Now if I hang up the real phone, the state remains unchanged on the side
of Asterisk. Both the D and B1-LEDs remain on.
Only after I hang up in the Softphone, more begins to happen in the log:
First it tells that the call was answered, then it talks about the
hangig up-process.
This is how a call looks:
-- Executing Dial(SIP/12346-457f, CAPI/0442607572:b012669095|30) in
new stack
-- creating pipe for PLCI=-1
sent CONNECT_REQ MN =0x4
-- Called 0442607572:b012669095
-- CAPI[contr1/0442607572]/0 answered SIP/12346-457f ---
-- CAPI Hangingup
sent DISCONNECT_B3_REQ NCCI=0x10101
sent DISCONNECT_REQ PLCI=0x101
I've marked the interesting line.
After begining to dial, the lines until Called 044... appear. Then
nothing happens besides the real phone actually ringing. Even if I
answer it, nothing happens in Asterisk or in X-Lite.
Then, when I hang up in X-Lite, the rest of above lines is printed.
If I don't answer the real phone, the line marked above is not printed.
The rest is the same.
So it's like Asterisk not getting a signal from the CAPI-layer that the
phone on the other side was actually answered.
What do I have to tweak? Which file do you actually need to help me?
I've included capi.conf and the relevant parts of extension.conf below
(as copied and pasted from various tutorials out there).
I'd gladly appriciate any help.
Philip
capi.conf:
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=0442607572
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=demo
devices=2
extension.conf:
[ch-fest-netz]
exten = _0[1-9].,1,Dial(CAPI/0442607572:b${EXTEN},30)
exten = _0[1-9].,2,Hangup
[theflintstones]
include = ch-fest-netz
exten = _[123456789],1,NoOp(call for ${EXTEN})
exten = _[123456789],2,Dial(SIP/${EXTEN},60,tr)
exten = _[123456789],3,Congestion
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Re: [Asterisk-Users] can a sip.conf stanza be shared by several phones?
If several phones register to the same sip.conf section what will happen with a Dial SIP/shared in asterisk? All phones ringing and the first one to answer gets the call? Undefined behavior? I believe the last one to register will be handed calls destined to that extension. If you want multiple phones to ring, then each phone should have its own unique registration, and your extensions.conf entry should look something like: 555,1,Dial(SIP/101SIP/102,15) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: gsm player for Linux?
Jesse Guardiani wrote: Hello, Does anyone know of an audio player for Linux that will play asterisk's GSM files? I know I can convert them using sox, but I'm hoping to play them natively so I can test out some voicemail settings. Scratch that. Looks like sox can play it. Thanks! -- Jesse Guardiani, Systems Administrator WingNET Internet Services, P.O. Box 2605 // Cleveland, TN 37320-2605 423-559-LINK (v) 423-559-5145 (f) http://www.wingnet.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] can a sip.conf stanza be shared by several phones?
Hi, Only one phone can be register to the same sip.conf section. Only the last registered phone will ring. You need to have 1 section for each phone. To have all phones ringing, you need to use (in extensiosn.conf) : Exten = 1,1,Dial(SIP/phone1SIP/Phone2SIP/Phone3) Regars, Fred. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Louis-David Mitterrand Envoyé : lundi 28 mars 2005 16:44 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] can a sip.conf stanza be shared by several phones? Hi, If several phones register to the same sip.conf section what will happen with a Dial SIP/shared in asterisk? All phones ringing and the first one to answer gets the call? Undefined behavior? Thanks, -- Jesus is coming! Everyone look busy! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP images
On Monday 28 March 2005 14:58, Tony Hoyle wrote: Ron Wellsted wrote: What route is left for guy with a few Cisco phones in Europe? Piracy? I looked around for nearly a year for a contract after a kind soul got me the images (the closest I got was a site in the US who were prepared to sell me the CON-SNT-CP7960 for £8 ... with £150 Postage!!!)... eventually gave up and ordered a CON-SNT-PKG1 package from lanway which I managed to get for £42. Of course being a Cisco contract it still hasn't arrived 2.5 weeks later. Cisco are the first company I've ever come across who seem to actively resent having customers and would rather you went with someone else. It doesn't arrive. It's all done instantly via email. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] can a sip.conf stanza be shared by several phones?
That last phone which registers will receive all the calls. This depends on the registration frequency set on the various phones, and will most likely be very unpredictable. If you want all phones to be usable, you need multiple SIP sections. -Original Message- From: Louis-David Mitterrand [mailto:[EMAIL PROTECTED] Sent: Monday, March 28, 2005 8:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] can a sip.conf stanza be shared by several phones? Hi, If several phones register to the same sip.conf section what will happen with a Dial SIP/shared in asterisk? All phones ringing and the first one to answer gets the call? Undefined behavior? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] another voipjet question
I'm working on it - I only started a week ago - and then I didn't know I
wanted to do all these other things with it. * is adictive!
Art Zemon wrote:
Tim Litwiller wrote:
so where did you put these lines?
exten = _1NXXNXX,1,SetCallerID(4153574000)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011.,1,SetCallerID(4153574000)
exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
Tim,
I did not use those lines. If you set up the two trunks as I described,
AAH will route calls out through voipjet. You don't have to manually add
those lines.
I want asterisk to use my pots line for local calls and voipjet for
long distance after the initial 100 free minutes my line provider give
with our plan. but to failover if one is busy and the other isn't.
Ahhh... *now* I think you need to get familiar with writing Asterisk
config files. :-)
-- Art Z.
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RE: [Asterisk-Users] Asterisk on a dialup connection?
This is what I get: speex - - - - - - - - - - - -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Monday, March 28, 2005 1:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk on a dialup connection? Kerry Garrison wrote: Firefly supports Speex too but trying it just now I am getting no audio. -Kerry Type show translation in your Asterisk console to check if you have speex installed. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on a dialup connection?
here is what i get speex -11 5 511 5 412 - -43 - Original Message - From: Kerry Garrison [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, March 28, 2005 10:13 AM Subject: RE: [Asterisk-Users] Asterisk on a dialup connection? This is what I get: speex - - - - - - - - - - - -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Monday, March 28, 2005 1:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk on a dialup connection? Kerry Garrison wrote: Firefly supports Speex too but trying it just now I am getting no audio. -Kerry Type show translation in your Asterisk console to check if you have speex installed. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI/Dialing out
Philip Hofstetter wrote:
capi.conf:
[..]
[interfaces]
msn=0442607572
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=demo
devices=2
extension.conf:
[ch-fest-netz]
exten = _0[1-9].,1,Dial(CAPI/0442607572:b${EXTEN},30)
Are you sure 044260xxx is your MSN? In germany the MSN is your phone
number without the local area code.
rgds
pos
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Re: [Asterisk-Users] Problem parsing unusual SIP/SDP
Stewart Nelson wrote: Well, provider is now sending a different tag, so Asterisk does not find a match, assumes that this response is for a call it does not know about, and discards it. Yes, that is what is happening here. That makes sense, but since Asterisk always generates a unique Call-ID for each call, I would think that tag checking on outgoing calls would be unnecessary. However, the routine carefully chooses the From or To field according to the call direction, so there is probably a good reason to check all calls. Indeed, the change that I would request might break operation with some other provider or device. Right. Strictly speaking, we should always be checking tags, and in this case (where the SIP provider is doing the correct thing but we are not) then Asterisk should be fixed to do it properly, rather than just avoiding the tag checking. Is it worth posting such a vague bug report? Unfortunately, I know absolutely nothing about the internals of Asterisk. Yes, please do, but make sure you include a full 'sip debug/set verbose 255/set debug 255' as an attachment in the bug. Also include the relevant portions of your sip.conf file (with secrets removed, of course). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pass caller ID to another application or machine.
--- Michiel van Baak [EMAIL PROTECTED] wrote: I created that file myself. That way I can put debug information into that logfile while developing that agi script. It's part of my skeleton agi script ;) Please pardon my ignorance, but how did you get asterisk to pass that into to your log file. That is in essence the part I'm haveing the most difficulty with. Thanks, Richard __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI/Dialing out
Philip Hofstetter wrote: msn=0442607572 incomingmsn=* There's already been a suggestion to drop your area code. That may or may not work in Germany as I don't know how MSNs are presented. In Holland I had to have msn=201234567 Where the number would normally be quoted as 0201234567, ie dropping the 0. This gets corrected on called id from /etc/asterisk/capi.conf's [general] section which reads as follows: [general] nationalprefix=0 internationalprefix=00 ... Dunno if this will work for you but it all works fine for me. cw ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP images
Bob Goddard wrote: It doesn't arrive. It's all done instantly via email. There's a whole package apparently (hence the £150 postage I was quoted, although I suspect they just weren't interested in selling). Even the entry on voip-info.org says it takes two weeks... Once you buy it the request goes to Cisco who have to get off their backsides and actually issue you with the thing. Nothing yet, although I'll be chasing it again tomorrow (unfortunately it's impossible to chase it directly with cisco as they refuse to deal with mere customers). I've come *so* close to putting the phone on ebay and forgetting about it. Certainly I'll never buy a cisco product again. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting quadbri to EuroISDN with 2 TE and 2 NT ports - what cables and settings ?
the cables are all straight cables but it seems you did not set the jumpers on port4 On Mon, 28 Mar 2005 16:34:18 +0200, Robert Rozman [EMAIL PROTECTED] wrote: Hi, I'm trying to connect quadbri between powered ISDN phone and ISDN line: ISDN ---1--- TE - * - NT --2-- Phone I use quadbri, suse 9.2 and latest 0.2.0-RC7k bristuff. I've used sample settings provided with package, but do get strange error (I think that I have wrong setting for P2P or P2MP setting and cables 1 and 2). If I connect phone to ISDN with straight cable it works. I've put quadbri in between, and connected ISDN to span1 in TE mode, and phone in NT mode on span4. Did configuration (added at the end). I get errors: qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 1 I'm not sure what cables to use. I use straight for -1- and -2-. Is this right ? I'm in European community (EuroISDN) so I guess I should have some pretty standard connections... I have ISDN line with 2 MSNs and two connectors on NT termination - so I guess this is P2MP. Is this OK ? Does anyone have working example for Germany ? There is also one strange thing: I get this in dmesg when loading qozap module, although I have specified 4th port to be NT - and pri show span 4 shows it in netowork mode: Zapata Telephony Interface Unloaded module zaptel unsupported by SUSE/Novell, tainting kernel. Zapata Telephony Interface Registered on major 196 module qozap unsupported by SUSE/Novell, tainting kernel. PCI: Enabling device :02:0c.0 ( - 0003) ACPI: PCI interrupt :02:0c.0[A] - GSI 20 (level, low) - IRQ 209 qozap: S/T ports: 4 [ TE TE TE TE ] qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Registered tone zone 3 (Netherlands) Thanks in advance, regards, Rob. #- /etc/zaptel.conf: loadzone=nl defaultzone=nl # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 #- /etc/asterisk/zapata.conf [channels] switchtype = euroisdn pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres=yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 ;--- ; p2p TE mode (for connecting ISDN lines in point-to-point mode) ;signalling = bri_cpe ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp context=isdn-incoming group = 1 ; S/T port 1-3 (first quadBRI, or lower ports of an octoBRI) channel = 1-2 channel = 4-5 channel = 7-8 ;--- ; p2p NT mode (for connecting an ISDN PBX in point-to-point mode) signalling = bri_net context=pbx-incoming group = 2 ; S/T port 4 (second quadBRI, or upper ports of an octoBRI) channel = 10-11 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michal Bielicki http://www.asterisk.com.pl/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which analog phones to use and why?
Hello! Now that I finally have my TDM board working, I want to move forward with using PBX functions. However, it seems cumbersome to use standard POTS telephones with Asterisk. I know that there are many of you installing even large systems based on channel banks and analog telephones. What phones are you using? How do you simulate phone system features on a phone that doesn't have extra buttons? Or are you all using ADSI telephones? It seems that for the price of a ADSI telephone (never mind the cost per channel of a channel bank and T1 card), you can get a good quality IP telephone. In that case, what is the appeal of analog? Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pass caller ID to another application or machine.
On 07:50, Mon 28 Mar 05, Richard Reina wrote: Please pardon my ignorance, but how did you get asterisk to pass that into to your log file. That is in essence the part I'm haveing the most difficulty with. Richard, I did this (as root) touch /var/log/asterisk/my_agi.log chmod 777 /var/log/asterisk/my_agi.log that way the logfile is there. now you can use this file to log from inside your agi script: if ($debug) fputs($stdlog, read: $input\n); All set and ready to go :) -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 and multiple simultaneous calls
Hate to bring up an old thread. I just configured a 7960 with multiple lines appearing. Each defined the same, but the buttons don't seem to roll over. What else do I have to define to do this. Henry - Original Message - From: Chris Wade [EMAIL PROTECTED] To: C F [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 23, 2005 1:12 PM Subject: Re: [Asterisk-Users] Cisco 7940 and multiple simultaneous calls C F wrote: These phones simply realize that the account information for both lines is the same and assumes (possibly incorrectly) that it should accept calls to that 'account' on all lines with that account. possibly incorrectly I think you are incorrect, this has the *best* result, it will just ring the second line when a second call comes in. I'm just saying that the phone should be 'dumb' and just do what it is told to do instead of assuming anything. However, in the case of the cisco phones, there isn't a config option of which way it should work - assuming it is, after all, the phone doing all this. Either way, I agree that this is the *best* solution if you really need that many calls. This disables the phones ability to handle two incoming calls per line button however. Maybe, I use it with disabling call waiting, so I dont realy know if this is true, in any case besides for disabling or not disabling the second call on the same line, it will also rollover to the next button, but this is the desired result in most cases, other wise why don't you use a different sip account for the next button, if you are using the same sip account, then you want the second call to go to the next button (unless you want 12 calls on a Cisco 7960?). If my memory serves me correctly, what I described is indeed the way it works. As for your last statement/question, some people do want 12 incoming calls on the 7960. My work-around, and I'm sure many others too, was to create a -a and -b 'account' for each 'account' and then do dialplan rollover to make the 7940 accept two calls per line button, or 4 simultaneous incoming calls. The 7960 could accept 12 simultaneous incoming calls this way using a -a through a -f 'account'. Correct, but it's usualy much better and easier to have the phone handle the rollover, who needs more than 6 simultanewous calls? whats wrong with call parking if you do need more than 6? It's up to the individual installation as to which method is better. I've setup both before and even the individual station user typically had a preference of which way it worked. I've yet to see somebody actually need even 6, but who knows. And nothings wrong with parking, just didn't bring it up cause it was somewhat OT. -Chris PS: Haven't checked this, but the phone may actually register per 'line' meaning it would register multiple times, but since ALL the details of the register are the same, * just treats it as a re-register and neither * nor the phone know the difference, so both 'work together' to produce this effect. It's possible, since it comes from the same IP address asterisk knows that it's the same and doesn't give any errors (I never tried with sip debug, so I don't know), but I think it's actualy the phone that does it, in any case it doesn't matter, as long as it works. Same here, haven't done a debug to see what is actually happening, but it just works :) -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: * - SMS w/out PSTN
Chris, I'm guessing you were one of our beta testers migrated over to a live account. This may be a configuration issue on your account. Best to contact [EMAIL PROTECTED] - be sure to tell them your account name (email address you registered with). Kind regards, -- Adam Holt Bayham Systems Ltd Web:http://www.bayhamsystems.com/asterisk.html Email:adam.holt at bayhamsystems.co.uk Address: No. 1 Farnham Road, Guildford, Surrey, GU2 4RG, United Kingdom *** NEWS: Bayham Systems delivers first ever Global SMS Auction for UNICEF / GSM-Association. See how this raised over EUR90,000 at: http://www.gsm.org/ -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Chris HARIGA Sent: Thursday, March 24, 2005 4:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] * - SMS w/out PSTN I have one account with 65 credits at http://www.bayhamsystems.com/ and I send 2 messages to my Sprint cell. Today, after 3 days, I'm still waiting for those messages. If someone makes it work please let me know. I try first message to send thru * and the second directly form the website form. On the web, at SMS status I have success on all :) Best regards, Chris HARIGA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP images
If you call Cisco contract support. 1-800-447-9347 and give them the serial number used when you purchased the smartnet they will give you the contract number over the phone. If the contract was sold properly the reseller would have asked you for the serial number of the unit and turned that into Cisco. Cisco should have then emailed the contract number to you. My experience has been they only email you about half the time and you have to call them the other half. Henry - Original Message - From: Tony Hoyle [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 28, 2005 10:06 AM Subject: Re: [Asterisk-Users] Cisco 7960 SIP images Bob Goddard wrote: It doesn't arrive. It's all done instantly via email. There's a whole package apparently (hence the £150 postage I was quoted, although I suspect they just weren't interested in selling). Even the entry on voip-info.org says it takes two weeks... Once you buy it the request goes to Cisco who have to get off their backsides and actually issue you with the thing. Nothing yet, although I'll be chasing it again tomorrow (unfortunately it's impossible to chase it directly with cisco as they refuse to deal with mere customers). I've come *so* close to putting the phone on ebay and forgetting about it. Certainly I'll never buy a cisco product again. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI STREAM FILE command
On Mon, 2005-03-28 at 08:30 -0600, Bill Kervaski wrote: Has anyone had success with the AGI STREAM FILE command with the CVS? I can't get it to work with the debian 1.0.5 package or the CVS on Redhat or Debian. It's not syntax, I'm doing that right. It doesn't give me an error when I use AGI DEBUG, it doesn't even give a response, just goes right on to the next command. I put a SAY NUMBER 123 # before and after the STREAM FILE and they both work fine, returning 200 OK, etc. DO NOT SEND A DIGEST TO THE MAILING LIST You do not mention if you followed the suggestion someone else made about making sure you do not have the extension on the file. Without providing the line you are trying to make work, you can not make most of us believe you haven't made a mistake. There are way too many people using Asterisk AGI successfully for it to likely be a bug in the Asterisk code if your command is simple. With as much as I love Debian, the distribution is not the problem. Do not use the very old Debian asterisk packages though. They are so old as to have well known bugs. If you want to participate better in the mailing list, maybe you would be better off to remove the digest option from the mailing list and use a proper mail filter to split the list mail to a folder other than your inbox. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AMP-1.10.007 Released!
Hello all, The Secret Agent final release of the Asterisk Management Portal is now available for download: http://amp.coalescentsystems.ca/ This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback! 1.10.007 - Added AMP Users (multi-department, basic multi-tenant) - Added incremental upgrade script (install_amp) - Use /etc/amportal.conf to tweak AMP to your environement (MySql credentials, web root, ip address, etc) - New Outbound Routes page to control trunks used for outbound calls based on dial patterns - LCR using Outbound Routes - Trunks page adds dial rules to modify numbers per-trunk before dialing - ENUM Trunks - Queues support added - Support for ZAP extensions - More voicemail options added - New AGI-based directory application to support both first and last name lookups and return to operator - provide customization points for all AMP generated extension contexts. - Upgrade to Flash Operator Panel 0.20 - Upgrade Asterisk-Stat to v2.0 - Added cvs2cl generated ChangeLog (see this for all changes and bug fixes) Regards, Ryan ___ Ryan CourtnageCoalescent Systems Inc Director CTO Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which analog phones to use and why?
On Mon, 2005-03-28 at 12:29 -0500, [EMAIL PROTECTED] wrote: Hello! Now that I finally have my TDM board working, I want to move forward with using PBX functions. However, it seems cumbersome to use standard POTS telephones with Asterisk. I know that there are many of you installing even large systems based on channel banks and analog telephones. What phones are you using? How do you simulate phone system features on a phone that doesn't have extra buttons? Or are you all using ADSI telephones? It seems that for the price of a ADSI telephone (never mind the cost per channel of a channel bank and T1 card), you can get a good quality IP telephone. In that case, what is the appeal of analog? Depends on what functions you are trying to implement. Hold isn't hard on a regular phone. Transfer isn't hard. Voicemail access isn't hard. Beyond that, there isn't a lot that needs to be done. If you find that you need more functions, then you may need to move up to a SIP phone. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bristuff-0.2.0-RC7k: error on loading qozap : qozap: Unknown symbol zt_xxxxx
Hi, I had problems described in another thread so went from a start, but now have problems when loading qozap module. I get : # insmod qozap.ko ports=9 insmod: error inserting 'qozap.ko': -1 Unknown symbol in module and in /var/log/messages: module qozap unsupported by SUSE/Novell, tainting kernel. qozap: disagrees about version of symbol zt_receive qozap: Unknown symbol zt_receive qozap: disagrees about version of symbol zt_ec_chunk qozap: Unknown symbol zt_ec_chunk qozap: disagrees about version of symbol zt_transmit qozap: Unknown symbol zt_transmit qozap: disagrees about version of symbol zt_unregister qozap: Unknown symbol zt_unregister qozap: disagrees about version of symbol zt_register qozap: Unknown symbol zt_register I did start from clean Suse 9.2 : cd /usr/src/linux make clean make mrproper make cloneconfig make prepare-all ln -s /usr/src/linux-2.6.8-24.13/ /usr/src/linux-2.6 cp /usr/src/linux-2.6.8-24.13-obj/i386/smp/Module.symvers /usr/src/linux // cause of warning when compiling zaptel echo # Section for zaptel device /etc/udev/rules.d/50-udev.rules echo KERNEL=\zapctl\, NAME=\zap/ctl\ /etc/udev/rules.d/50-udev.rules echo KERNEL=\zaptimer\, NAME=\zap/timer\ /etc/udev/rules.d/50-udev.rules echo KERNEL=\zapchannel\, NAME=\zap/channel\ /etc/udev/rules.d/50-udev.rules echo KERNEL=\zappseudo\, NAME=\zap/pseudo\ /etc/udev/rules.d/50-udev.rules echo KERNEL=\zap[0-9]*\, NAME=\zap/%n\ /etc/udev/rules.d/50-udev.rules echo /etc/udev/rules.d/50-udev.rules echo zap/*:root:root:660 /etc/udev/permissions.d/50-udev.permissions echo /etc/udev/rules.d/50-udev.rules cd zaphfc/ wget http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC7k.tar.gz tar zxvf bristuff-0.2.0-RC7k.tar.gz cd bristuff-0.2.0-RC7k/ ./download.sh ./compile.sh #clearing SuSE deprecated SuSE modules for module in /lib/modules/`uname -r`/misc/*; do rm -i /lib/modules/`uname -r`/extra/$(basename $module); done #Loading the drivers (quadBRI): cd qozap modprobe zaptel #insmod qozap.o (for kernel 2.4) insmod qozap.ko ports=9 (for kernel 2.6) //This is where I got errors... ztcfg Any advice, what's wrong ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remove a channel from receiving inbound calls
Ive got a small office setup with a TDM400 and 3 FXO cards. Id like to take away the ability of the 3rd FXO to receive calls (as this line runs through our old-fashioned fax machine) BUT still be able to use it for outbound calls. With our original, and very basic PBX we could modify the auto attendant on a particular PSTN line to pick up after 4 rings (which would allow the fax to pickup after 2) while the rest of the lines picked up after 1 ring. Anyone have a simple way for me to do this? Bob Sowers Oak Leaf Systems Network Engineer 207.498.2510 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Third party Firefly issue very weird??
When I connect to the third party softphone (firefly) I get connected at my house and at my office where I have the asterisk..but when I went to my friends house to set him up his firefly showed a gray circle like it was not connecting at all? Has Anyone seen this happen what is causing this no to connect, does anyone know ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Third party Firefly issue very weird??
First guess.. firewall. Jon Walsh wrote: When I connect to the third party softphone (firefly) I get connected at my house and at my office where I have the asterisk..but when I went to my friends house to set him up his firefly showed a gray circle like it was not connecting at all? Has Anyone seen this happen what is causing this no to connect, does anyone know ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP-1.10.007 Released!
How does one downlaod the upgrade only is there the ability to do so from the software or do you need to re-burn an iso or is the iso an upgrade version or the whole install over again? Jonathan On Mon, 28 Mar 2005 09:39:40 -0700, Ryan Courtnage [EMAIL PROTECTED] wrote: Hello all, The Secret Agent final release of the Asterisk Management Portal is now available for download: http://amp.coalescentsystems.ca/ This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback! 1.10.007 - Added AMP Users (multi-department, basic multi-tenant) - Added incremental upgrade script (install_amp) - Use /etc/amportal.conf to tweak AMP to your environement (MySql credentials, web root, ip address, etc) - New Outbound Routes page to control trunks used for outbound calls based on dial patterns - LCR using Outbound Routes - Trunks page adds dial rules to modify numbers per-trunk before dialing - ENUM Trunks - Queues support added - Support for ZAP extensions - More voicemail options added - New AGI-based directory application to support both first and last name lookups and return to operator - provide customization points for all AMP generated extension contexts. - Upgrade to Flash Operator Panel 0.20 - Upgrade Asterisk-Stat to v2.0 - Added cvs2cl generated ChangeLog (see this for all changes and bug fixes) Regards, Ryan ___ Ryan CourtnageCoalescent Systems Inc Director CTO Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP images
It doesn't arrive. It's all done instantly via email. There's a whole package apparently (hence the £150 postage I was quoted, although I suspect they just weren't interested in selling). Even the entry on voip-info.org says it takes two weeks... Once you buy it the request goes to Cisco who have to get off their backsides and actually issue you with the thing. Nothing yet, although I'll be chasing it again tomorrow (unfortunately it's impossible to chase it directly with cisco as they refuse to deal with mere customers). I've come *so* close to putting the phone on ebay and forgetting about it. Certainly I'll never buy a cisco product again. As a side note to the above (in the US), the contract reseller is suppose to obtain the phone's serial number. If that serial number is not registered to the individual requesting the contract, the contract supposedly will not be issued. That process is apparently used to identify when used phones are sold via eBay (etc), and essentially says one does not have a valid software license therefore it cannot be placed on maintenance. (A software license cannot be transferred with the sale of a used phone or any of cisco's equipment.) That same process is used for all Cisco equipment, however some used equipment resellers have been able to find ways around it (one way or another). Once a maintenance contract number has been issued (regardless of whether its on a piece of paper or email), that contract number has to be entered into a cisco system that tracks the number against a customer account. If you don't have a customer account, that process can't be completed either. Some resellers will create your account for you and others won't. Once the account has been created and the contract recorded, then the customer is granted access to the download sections of their site via their login/authentication process. So the bottom line is the process requires a fair amount of manual labor and for $8 (in the US), few resellers have any interest in the sales commission resulting from an $8 sale. (Guess that says if you're buying 500 contracts, one might receive a different level of reseller interest.) Regardless of whether we like it or not, cisco wrote the license terms and asterisk users are not going to change their machine. It's obviously written to discourage reselling used equipment without paying a re-certification fee, and that re-certification re-license process can get to be far more costly then simply purchasing their new equipment. Surprise surprise! I don't work for cisco or any of their resellers. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP-1.10.007 Released!
On Monday 28 March 2005 12:19, Jon Walsh wrote: How does one downlaod the upgrade only is there the ability to do so from the software or do you need to re-burn an iso or is the iso an upgrade version or the whole install over again? Jonathan You can do it manually, or through a script like this one: wget szmidt.org/asterisk/asterisk-update.sh There's a line which let's you specify a specific release like 1-0-7. You can get both the developer version or stable. Don't forget to chmod 700 /usr/local/sbin/asterisk-update.sh -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP-1.10.007 Released!
On Mon, 28 Mar 2005 12:24:00 -0500 steve szmidt [EMAIL PROTECTED] wrote: On Monday 28 March 2005 12:19, Jon Walsh wrote: How does one downlaod the upgrade only is there the ability to do so from the software or do you need to re-burn an iso or is the iso an upgrade version or the whole install over again? Jonathan You can do it manually, or through a script like this one: wget szmidt.org/asterisk/asterisk-update.sh There's a line which let's you specify a specific release like 1-0-7. You can get both the developer version or stable. Don't forget to chmod 700 /usr/local/sbin/asterisk-update.sh -- Steve Szmidt Steve, I believe you have confused AMP-1.10.007 with ASterisk 1.0.7. This was an inquiry about AMP and not Asterisk. Two different beasts. Jon, The AMP program is an addon to ASterisk. It is not in ISO form. If you go to the AMP web site, http://amp.coalescentsystems.ca/ and download the tar file, you can uncompress it and do an upgrade. The upgrade instructions were included in the original AMP announcement... Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP-1.10.007 Released!
On Monday 28 March 2005 12:32, Robert Webb wrote: Steve, I believe you have confused AMP-1.10.007 with ASterisk 1.0.7. This was an inquiry about AMP and not Asterisk. Two different beasts. Hehe, I do believe you are right! Thanks! Maybe I should practice my reading... : ) -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remove a channel from receiving inbound calls
On Mon, 2005-03-28 at 12:02 -0500, Bob Sowers wrote: Ive got a small office setup with a TDM400 and 3 FXO cards. Id like to take away the ability of the 3rd FXO to receive calls (as this line runs through our old-fashioned fax machine) BUT still be able to use it for outbound calls. With our original, and very basic PBX we could modify the auto attendant on a particular PSTN line to pick up after 4 rings (which would allow the fax to pickup after 2) while the rest of the lines picked up after 1 ring. Anyone have a simple way for me to do this? No need to go to all that trouble. Zap can be set up to automatically detect fax tones. Then you can configure incoming faxes to be directed to you fax machine and do whatever you want with everything else. Gone are the days of dedicated fax lines. http://www.voip-info.org/tiki-index.php?page=Asterisk%20fax Look at the faxdetect stuff. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP-1.10.007 Released!
Hi Jon, On 28-Mar-05, at 10:19 AM, Jon Walsh wrote: How does one downlaod the upgrade only is there the ability to do so from the software or do you need to re-burn an iso or is the iso an upgrade version or the whole install over again? Jonathan If you have already installed a previous version of Asterisk Management Portal, just download the AMP-1.10.007.tar.gz and run the commands listed in the 'UPGRADE' document. The new upgrade script will ensure a smooth transition to the latest version. Ryan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to config speex?
Hello, I'm using Asterisk 1.0.7 and speex codec, but can not found the codec.conf file to change the setting of speex. Does anybody know how to change the bit rate, quality..etcof speex? Thanks! BR, Dominic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP-1.10.007 Released!
Okay thanks Lads but I am completely afraid that if I unpak this file in the directory of AMP will it override my current settings and I haven't found where you saying the upgrade instructions are with teh original announcement?? I am a newbie can you tell On Mon, 28 Mar 2005 09:39:40 -0700, Ryan Courtnage [EMAIL PROTECTED] wrote: Hello all, The Secret Agent final release of the Asterisk Management Portal is now available for download: http://amp.coalescentsystems.ca/ This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback! 1.10.007 - Added AMP Users (multi-department, basic multi-tenant) - Added incremental upgrade script (install_amp) - Use /etc/amportal.conf to tweak AMP to your environement (MySql credentials, web root, ip address, etc) - New Outbound Routes page to control trunks used for outbound calls based on dial patterns - LCR using Outbound Routes - Trunks page adds dial rules to modify numbers per-trunk before dialing - ENUM Trunks - Queues support added - Support for ZAP extensions - More voicemail options added - New AGI-based directory application to support both first and last name lookups and return to operator - provide customization points for all AMP generated extension contexts. - Upgrade to Flash Operator Panel 0.20 - Upgrade Asterisk-Stat to v2.0 - Added cvs2cl generated ChangeLog (see this for all changes and bug fixes) Regards, Ryan ___ Ryan CourtnageCoalescent Systems Inc Director CTO Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which analog phones to use and why?
Steven Critchfield [EMAIL PROTECTED] wrote on 03/28/2005 11:44:03 AM: Depends on what functions you are trying to implement. Hold isn't hard on a regular phone. Transfer isn't hard. Voicemail access isn't hard. Beyond that, there isn't a lot that needs to be done. If you find that you need more functions, then you may need to move up to a SIP phone. Well, what it seems to come down to is two things: 1) People *expect* business phones to just plain have more buttons 2) People want one-button convenience For example, people want to be able to push a single button to reach at least a selection of internal extensions. Or, they want to be able to press a single button for parking a call, or voicemail, or who-knows-what. Of course, a standard analog phone can't do those things: it doesn't have the buttons! :) I guess even a telephone with speed dial buttons could do that, maybe? Something like this: http://www.101phones.com/flypage/2126/8a3a9cb7ed9a26e52f4129070e30b829/Panasonic_KX-TS105W I was just wondering how others are addressing this. You can't all be making receptionists memorize codes, are you? :) Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AMP-1.10.007 Released!
Ryan, -Original Message- If you have already installed a previous version of Asterisk Management Portal, just download the AMP-1.10.007.tar.gz and run the commands listed in the 'UPGRADE' document. The new upgrade script will ensure a smooth transition to the latest version. /Snip/ How close is Coalescent relaeasing a version that is self installable on a new system as a first time install, assuming the guys are using Standard flavors like Redhat or Debian? I understand the complexities involved in different flavors, but Redhat being the most used, is there any plan to release a build that self installs on Redhat flavors? Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP images
Henry Devito wrote: If you call Cisco contract support. 1-800-447-9347 and give them the serial number used when you purchased the smartnet they will give you the contract number over the phone. If the contract was sold properly No serial number was asked for.. I just explained that I just wanted the smartnet contract and they took my credit card details. Presumably not all dealers work the way cisco would like them to. TBH I'm not even sure I know the serial of that phone - threw the box away months ago. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] need to install the openline4 card
Dear All, im new to asterisk, im so much intrested in this system, i have purchased openline4 card and tried to install it in my redhat linux 9.0 machine, i have followed up with all the steps in the installation file for this card but it didnt work with me, can you please help to install it in my linux box in order to start using the asterisk. ?! Best Regards, Alex Do you Yahoo!? Yahoo! Small Business - Try our new resources site! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP-1.10.007 Released!
On 28-Mar-05, at 10:48 AM, Jon Walsh wrote: Okay thanks Lads but I am completely afraid that if I unpak this file in the directory of AMP will it override my current settings and I haven't found where you saying the upgrade instructions are with teh original announcement?? I am a newbie can you tell Please use the project mailing lists for question regarding AMP: http://sourceforge.net/mail/?group_id=121515 #cat /usr/src/AMP/UPGRADE Upgrading from the AMP-version.tar.gz download from sourceforge: - save the AMP-version.tar.gz to /usr/src/ - cd /usr/src/ :: change to your /usr/src directory - rm -rf AMP :: remove your current AMP directory - tar -zxvf AMP-version.tar.gz :: extract the new AMP tar - cd AMP :: change to the AMP directory - ./install_amp :: run the install script Ryan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP-1.10.007 Released!
[EMAIL PROTECTED] - Original Message - From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 28, 2005 12:51 PM Subject: RE: [Asterisk-Users] AMP-1.10.007 Released! Ryan, -Original Message- If you have already installed a previous version of Asterisk Management Portal, just download the AMP-1.10.007.tar.gz and run the commands listed in the 'UPGRADE' document. The new upgrade script will ensure a smooth transition to the latest version. /Snip/ How close is Coalescent relaeasing a version that is self installable on a new system as a first time install, assuming the guys are using Standard flavors like Redhat or Debian? I understand the complexities involved in different flavors, but Redhat being the most used, is there any plan to release a build that self installs on Redhat flavors? Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with 1 dialing (recording says must dial 1 when I thought I did)
TRUNKMSD1=1 ; MSD digits to strip
(usually 1 or 0)
TRUNKMSD2=2 ; MSD digits to strip
(usually 1 or 0)
; logn distance calls
exten = _91NXXNXX,1,NoOp(Dialing: ${TRUNK}/${EXTEN:${TRUNKMSD1}})
exten = _91NXXNXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD1}})
exten = _91NXXNXX,3,Congestion
When I dial a long distance number (916503270309 for example) I get the
message (I think from SBC) saying I must first dial a 1. Other times,
it works, like when I dial this number (914082341389).
Any ideas why would be appreciated.
Thanks,
-Peter
I have a TDM400P with two FXS and two FXO's.
My extensions.conf
TRUNKMSD1=1 ; MSD digits to strip
(usually 1 or 0)
TRUNKMSD2=2 ; MSD digits to strip
(usually 1 or 0)
; logn distance calls
exten = _91NXXNXX,1,NoOp(Dialing: ${TRUNK}/${EXTEN:${TRUNKMSD1}})
exten = _91NXXNXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD1}})
exten = _91NXXNXX,3,Congestion
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[Asterisk-Users] voicemail sending blank .WAV file via email
Ive recently installed asterisk and am working with the email a voicemail function. When a voice msg is left 4 files are created in the /var/spool directory. They are .gsm, .txt, .wav and .WAV. The .wav (lower case) has the actual audio in it, the .WAV is a short blank audio file. When * emails the message it is sending the .WAV and not the .wav file. Any thoughts would be appreciated. Jim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with 1 dialing (recording says must dial 1 when I thought I did)
Maybe the first digit is dialed before the dialtone, try adding a 'w'
before ${EXTEN..., e.g.
exten = _91NXXNXX,2,Dial(${TRUNK}/w${EXTEN:${TRUNKMSD1}})
Julian J. M.
On Mon, 28 Mar 2005 13:19:03 -0500, Kellner, Peter
[EMAIL PROTECTED] wrote:
When I dial a long distance number (916503270309 for example) I get the
message (I think from SBC) saying I must first dial a 1. Other times,
it works, like when I dial this number (914082341389).
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 8, Issue 229
On Sun, 27 Mar 2005, Nenad Radosavljevic wrote: Only way I have managed to get Zap channel to reject a call on TE110P without answering it, is to dial number that is not handled in dialplan (I have a ISDN PRI with 100 number DID service, and about 30 of them are handled by dialplan). So far I didn't manage to reject call that are handled in dialplan, except by Congestion command which answers the call first. There are two methods: * use the Hangup app after setting the PRI_CAUSE variable. This is the general way of sending a specified disconnect code. See http://www.voip-info.org/wiki-Asterisk+variable+PRI_CAUSE Tried this with the Stable 1.0.6 connected to Panasonic D500 and it indeed hung up a call from panasonics EXT or CO line but it doesn't give a busy or concestion signal to a caller (no sound to caller). I presume this is a Panasonic issue :( . * set the configuration option priindication=oob. This will make busy and congestion send an isdn disconnect insteadof playing audio. I belive that I have tried to set priindication=outofband and then tried with Congestion() in dialplan but it had a same effect as setting PRI_CAUSE + Hangup() . I'll try with the priindication=oob tomorow and i'll post results if there is any success. Nenad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI and SIP PHones in Asterisk
Does anybody has a link for a step by step explanation on how dows MWI works in Asterisk with a SIP phone? I hacve added the mailbox line in SIP.conf but i got nothing :( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MWI and SIP PHones in Asterisk
Did you also include an entry in voicemail.conf? After that the most common mistake is referencing a bad context for your VM. As long as you have it right, it should work fine. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robson Ribeiro Sent: Monday, March 28, 2005 2:03 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] MWI and SIP PHones in Asterisk Does anybody has a link for a step by step explanation on how dows MWI works in Asterisk with a SIP phone? I hacve added the mailbox line in SIP.conf but i got nothing :( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with 1 dialing (recording says must dial 1 when I thought I did)
[EMAIL PROTECTED] wrote on 03/28/2005 01:19:03 PM:
TRUNKMSD1=1 ; MSD digits to strip
(usually 1 or 0)
TRUNKMSD2=2 ; MSD digits to strip
(usually 1 or 0)
; logn distance calls
exten = _91NXXNXX,1,NoOp(Dialing: ${TRUNK}/${EXTEN:${TRUNKMSD1}})
exten = _91NXXNXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD1}})
exten = _91NXXNXX,3,Congestion
Your dial command is stripping the one. That's what the ${EXTEN:1} part
does. So, yes, you are dialing the 1, but the dial command is stripping
it.
If you want to keep the one, use this:
exten = _91NXXNXX,2,Dial(${TRUNK}/${EXTEN})
When I dial a long distance number (916503270309 for example) I get the
message (I think from SBC) saying I must first dial a 1. Other times,
it works, like when I dial this number (914082341389).
I have no idea where you're located. Is it maybe that you have 10-digit
dialing and that the one that works is a local call, and therefore does
not need the 1?
Tim Massey
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Re: [Asterisk-Users] Cisco 7960 SIP images
Serial number is on the bottom of phone. Email me off list I will help. - Original Message - From: Tony Hoyle [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 28, 2005 12:02 PM Subject: Re: [Asterisk-Users] Cisco 7960 SIP images Henry Devito wrote: If you call Cisco contract support. 1-800-447-9347 and give them the serial number used when you purchased the smartnet they will give you the contract number over the phone. If the contract was sold properly No serial number was asked for.. I just explained that I just wanted the smartnet contract and they took my credit card details. Presumably not all dealers work the way cisco would like them to. TBH I'm not even sure I know the serial of that phone - threw the box away months ago. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to config speex?
Title: Message As far as I know speex is an adaptive codec, i.e. it will automatically adjust to the conditions and provide the best quality possible. Therefore, there should be no need to configure that manually. Could anyone correct me if I'm wrong? On the other hand, I was thinking about getting my Asterisk and SJPhone run with Speex. Does anyone have experience? What's the quality boost compared to iLBC? Ist is difficult to install? Can I install speex on win2000/XP and use it from any softphone? How do I add speex to SJPhone's list of codecs? Any help will be appreciated. Thanks a lot, Roman -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dominic LuSent: Montag, 28. März 2005 19:43To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] How to config speex? Hello, I'm using Asterisk 1.0.7 and speex codec, but can not found the codec.conf file to change the setting of speex. Does anybody know how to change the bit rate, quality..etcof speex? Thanks! BR, Dominic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Verizon ISDN
I'm looking to use Asterisk with Verizon ISDN centex service in the US. I'd be connecting to an NT1 so I'd need an S/T interface. Users would have SIP phones registered with Asterisk and sharing the ISDN lines. The only PCI ISDN card that will support ISDN signalling in the US seems to be the Eicon Diva Server cards. These are hard to find and very expensive ($2500 for the Quad card). Its unclear to me looking at all the posts if anyone has successfully used a cheaper PCI card in the US. Does anyone have this working and where did you get the card? thanks, Brian G. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Verizon ISDN
Brian G wrote: I'm looking to use Asterisk with Verizon ISDN centex service in the US. I'd be connecting to an NT1 so I'd need an S/T interface. Users would have SIP phones registered with Asterisk and sharing the ISDN lines. ISDN BRI interfacing into a PC is hard to do in the US... there are just not many (if any) cards available to do it. The few cards that are available only have firmware for Euro-ISDN, not NI-2 (and US BRI is _not_ the same as anywhere else in the world). There are other options, though: Adtran (and others) make boxes that can cross-convert multiple BRIs into a PRI, which could then be connected to Asterisk via a T-1 card. Not an inexpensive way to go, though. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] need to install the openline4 card
What are exactly the errors you are getting? Once Again wrote: Dear All, im new to asterisk, im so much intrested in this system, i have purchased openline4 card and tried to install it in my redhat linux 9.0 machine, i have followed up with all the steps in the installation file for this card but it didnt work with me, can you please help to install it in my linux box in order to start using the asterisk. ?! Best Regards, Alex Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://us.rd.yahoo.com/evt=31637/*http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Jorge Merlino Teledata Comunicaciones Canelones 2101 - Montevideo - Uruguay [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Click-to-Talk with Asterisk? = TACI
Hi Dean, For the Click-to-Talk working by using TACI (Trivial Asterisk Call-Generator Interface). But I found one strange thing that, on my Snom 220 display it shows Asterisk Asterisk when someone tries to make a call...? Any idea where I could find this Asterisk Asterisk parameter. Neel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Click-to-Talk with Asterisk? = TACI
Nitesh Divecha wrote: But I found one strange thing that, on my Snom 220 display it shows Asterisk Asterisk when someone tries to make a call...? Any idea where I could find this Asterisk Asterisk parameter. Those are the default when no CLID or CNAM have been supplied when the call was generated. You should be able to modify the tool that is creating the call to supply some sort of appropriate information. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk and XLite on same machine (OSX)?
Aldo Bergamini wrote: [EMAIL PROTECTED] is believed to have said: Dear all, I have tried to run an asterisk instance together with XLite on a single machine (a PowerBook). The intent is to take advantage of IAX connections to easily cross NATs while traveling. While the IAX setup proved 'easy', just having to fiddle a little with working configs at both sides, I did not succeed so far in getting XLite to connect to the local Asterisk server, AND be able to make a call. [...] Strangely enough I can obtain again the login of the softphone, but I still get a 'call not approved' for any dialed number. Activating sip debug peer does not show anything while dialing; and no error/message shows in the Diagnostic Window of XLite. I solved the problem! It's completely unrelated to Asterisk. XLite does not support multiple accounts: I must have misinterpreted the meaning of the list of different proxies that can be found under SIP. As I already had a setting for use inside the office LAN, I wanted to leave it untouched; therefore I was adding more configurations in the next configuration entry points (Proxy 1, Proxy 2, etc). As soon as I simply edited the first configuration I got online in no time. So for any reason XLite seems either to have a bug with multiple configurations or just not support more than one different extension. (ok to me: it's a free 'lite' version; I can't really complain!) In the end I will simply get into my office extension over IAX2. Fine enough... Why don't you just use an IAX softphone in the first place, instead of a SIP softphone, plus asterisk in the middle? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 8 channel fxo setup outgoing call problem
I have an eight channel fxo setup (2 TDM400P cards)
and I have them setup. Here are my configs:
Zaptel.conf:
fxsks=1-8loadzone=usdefaultzone=us
Zapata.conf:
[trunkgroups]
[channels]
musiconhold=defaultrxwink=300;
Atlas seems to use long (250ms)
winksusecallerid=yeshidecallerid=nocallerid=xxcallwaiting=yesbusydetect=nocallprogress=nousecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesechotraining=800rxgain=0.0txgain=0.0
group=0callgroup=1pickupgroup=1immediate=no;faxdetect=incomingfaxdetect=no
signalling=fxs_kscallerid=asreceivedcontext=from-pstnchannel=1-8
Now calling in is no problem, all the
channels pick up the incoming call just fine. However, for some reason,
calling out does not work.
When I dial out (I have a default trunk
setup) I get this on the console:
-- Executing
Macro("SIP/201-1d90", "dialout-default|636399xxx") in new
stack -- Executing GotoIf("SIP/201-1d90", "1?4") in new
stack -- Goto
(macro-dialout-default,s,4) -- Executing
GotoIf("SIP/201-1d90", "1?6") in new stack -- Goto
(macro-dialout-default,s,6) -- Executing
Dial("SIP/201-1d90", "ZAP/g0/6363997681") in new stack --
Called g0/636399 -- Zap/1-1 answered
SIP/201-1d90
But it doesn't answer, nothing rings (locally or on my cellphone, the test
number I'm calling out to) it just says "connected" on the local phone but its
not actually connecting. This problem is very weird because as of 5 hours
ago, I only had 1 TDM400P card and 1 FXO chip and I could make outgoing calls
with no problems at all.
P.S.
I have checked the phone line I'm testing this on and it is
fine.
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Re: [Asterisk-Users] problem with 1 dialing (recording says must dial 1 when I thought I did)
[EMAIL PROTECTED] wrote on 03/28/2005 03:24:50 PM:
[EMAIL PROTECTED] wrote on 03/28/2005 01:19:03 PM:
; logn distance calls
exten = _91NXXNXX,1,NoOp(Dialing:
${TRUNK}/${EXTEN:${TRUNKMSD1}})
exten = _91NXXNXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD1}})
exten = _91NXXNXX,3,Congestion
Your dial command is stripping the one. That's what the ${EXTEN:1} part
does. So, yes, you are dialing the 1, but the dial command is stripping
it.
No, your command is correct: you need to strip the 9. Sorry about that.
Time for more coffee! :)
Tim Massey
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RE: [Asterisk-Users] 8 channel fxo setup outgoing call problem
Do you have ATI FXO daughter cards. We had experienced similar problems. After replacing the ATI with Digium X100M Rev B daughter cards the system has been running fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Mike Flynn Sent: March 28, 2005 3:05 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 8 channel fxo setup outgoing call problem I have an eight channel fxo setup (2 TDM400P cards) and I have them setup. Here are my configs: Zaptel.conf: fxsks=1-8 loadzone=us defaultzone=us Zapata.conf: [trunkgroups] [channels] musiconhold=default rxwink=300; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callerid=xx callwaiting=yes busydetect=no callprogress=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=incoming faxdetect=no signalling=fxs_ks callerid=asreceived context=from-pstn channel=1-8 Now calling in is no problem, all the channels pick up the incoming call just fine. However, for some reason, calling out does not work. When I dial out (I have a default trunk setup) I get this on the console: -- Executing Macro(SIP/201-1d90, dialout-default|636399xxx) in new stack -- Executing GotoIf(SIP/201-1d90, 1?4) in new stack -- Goto (macro-dialout-default,s,4) -- Executing GotoIf(SIP/201-1d90, 1?6) in new stack -- Goto (macro-dialout-default,s,6) -- Executing Dial(SIP/201-1d90, ZAP/g0/6363997681) in new stack -- Called g0/636399 -- Zap/1-1 answered SIP/201-1d90 But it doesn't answer, nothing rings (locally or on my cellphone, the test number I'm calling out to) it just says connected on the local phone but its not actually connecting. This problem is very weird because as of 5 hours ago, I only had 1 TDM400P card and 1 FXO chip and I could make outgoing calls with no problems at all. P.S. I have checked the phone line I'm testing this on and it is fine. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rhino Channel Bank or ADIT 600
I don't know what you mean by talk. Cost difference: well this is how I calculated it, one the overall cost of having to get an asterisk box that needs more CPU, and 2 the actual cost of the cards to make it work. As well as taking into considiration that adit is telco grade equipment, and has very good performance. Option 1: An adit 600 loaded with 6 FXS cards = $800, Digium Quad t1 = $1500. Option 2: An adit 600 with 5 FXS cards, and 1 CMG02 card = $1500 (Max, you could get it for cheaper). Option 1 for 384 (8 * 48) analog ports: 8 * 800 + 2 * 1500 = $9,400. (around $24.50 per port) Option 2 for 360 (9 * 40) analog ports: 9 * 1500 = $13,500. ($37.50 per port) But option 2 can all go on one asterisk box, since it's all voip, and no transcoding has to take place (in fact it's like getting 9 more asterisk boxes to give you some more horsepower). With option 1 you will run into problems of using more than one digium card on one box, although it works, it is not recommended. Consider that a dual xeon system costs $3000 you will end up paying $7,700 for each pair of 4 adit boxes, and this doesn't even give you the options of adding telco t1s. In my opinion using the CMG cards will pay out in every single way when you are talking of anything more than 2 Adit boxes to a single system. Couple observations. Adit with CMG uses MGCP vs SIP. Not sure how extensive the * support is for this. I believe Carrier Access is working on a SIP release but not sure how complete it will be. CMG/CMG2 are nice cards. Also have a nice license builtin for G729. If you use though you will need a matching one for the * server. However there is also a call limit of 12, I think, for the CMG and 24 for the CMG2 cards. Depending on application you could easily exceed this. So take the 8 T1 capacity with a grain of salt, you will not be able to have that may calls up at a time. Also one of my favorite applications is connecting an Adit directly via T1 to a Digium card. Then using another T1 port on the Digium card to connect to PRi from PSTN. This is called a traditional TDM switch. No IP in the patch, no headaches. Great for such things as FAX and ALARM circuits which are very problematic with IP in the path. It is not always about up front costs but about capabilities and support costs down the line. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Push VLAN to Polycom via DHCP
On Mar 27, 2005, at 12:10 AM, Matt Darnell wrote: Has anyone been succesful pushing a VLAN setting to a Polycom phone via DHCP? Chicken or the egg! How can the Polycom reach the proper DHCP server if it is not on the correct VLAN? That's why Ciscos and Polycoms support CDP, so the CDP-capable switch can supply the correct voice VLAN. I 'assumed' the phone would reboot with the new VLAN setting and get a new IP address from the DHCP server on the phone VLAN - there would be two DHCP servers. I can't think of any other way to make it work with DHCP. If it isn't designed to work that way, why would they put the option in the DHCP section. -Matt I had always understood that they only supported VLAN discovery via CDP. But reading the 1.4 admin guide it says this... VLAN ID See 2.2.1.2.2 DHCP Menu on page 7 Special Case: Cisco Discovery Protocol (CDP)a overrides Local FLASH which overrides DHCP VLAN Discovery. a. Can be obtained from a connected Ethernet switch if the switch supports CDP. This seems to imply that DHCP can be used to spec a VLAN. I too would like to find a way to make this work. I also do not think the phone would need to reboot. I have noticed being able to change the VLAN and have the tag applied or not without the phone rebooting. Actually about the only thing I can do and not have it reboot:-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 8 channel fxo setup outgoing call
I'm not sure what you mean, I had one card with one FXO in it and it worked fine. Now, we have 2 TDM400P cards with all of their slots filled with FOX chips and I can't make out going calls. Those are the only 2 cards on the system, the rest is just motherboard. -Original Message- Do you have ATI FXO daughter cards. We had experienced similar problems.After replacing the ATI with Digium X100M Rev B daughter cards the systemhas been running fine. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
