[asterisk-users] Queue Answer
Hi this is my setup: Customer - PRI - Server A with G729 - IAX2 Trunk(G729) - Server B - SIP Exten allowed codec=g729 - Snom phone Agents setup is working fine. I want when my agents are not available (queue) like not logged in or all are busy so no calls should come to my server b from server a I want my server a to not forward that call to my server b. Please guide me. Ive configured all my queue, sip exten on server b. server a is doing the routing of incoming calls to server b. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue and voicemail
Josué Conti wrote: Hi all, good? I would like to know if the option exists to together integrate the function of queue with the voicemail of the agent, or the pilot of the group. For example, in case that none of the agents of queue obtains to take care of a call, this call would be directed for a voicemail. I think you can do this in the dialplan - set a timeout for the queue, then route to voicemail when it expires. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation
Tom Rymes wrote: I dunno, I guess I'm not your mother, but then again, it seemed pretty rude considering the guy offered the program for free and you were criticizing the fact that he didn't develop a free linux app for you, too. Not specifically directed at Toms reply - Gee, all Stephen said was Ugh. This is a Win32 app, isn't it? ... that's hardly an opinion, at most perhaps a rethorical question. How was he being rude? I happen to agree that the tool discussed is of little value (as we are a Linux-only shop), but if Ugh. This is a Win32 app, isn't it? is enough to upset you, you really need to get out more. /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue and voicemail
You could alternatively set a context for your queue in your config and create an extension for voicemail, if you would rather give the option to go to voice mail to the caller... (example: They can dial 0 to leave a message) On 5/4/07, Per Jessen [EMAIL PROTECTED] wrote: Josué Conti wrote: Hi all, good? I would like to know if the option exists to together integrate the function of queue with the voicemail of the agent, or the pilot of the group. For example, in case that none of the agents of queue obtains to take care of a call, this call would be directed for a voicemail. I think you can do this in the dialplan - set a timeout for the queue, then route to voicemail when it expires. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RXFAX/TXFAX
Cesar Benjamin Garcia Martinez wrote: Somebody can tell me, what way i can send/receive faxes with asterisk 1.4??? [snip] How to i can send/receive fax to/from PSTN on asterisk 1.4 ? Check out a very recent thread on just that subject. Or go study how to use iaxmodem and hylafax. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04
Tzafrir; Actually I have found this config to work really well. I prefer to use a script run from inittab but Ubuntu doesn't work like Redhat or BSD. On a production box keeping asterisk up and running is THE TOP priority. If you would rather check every five minutes then replace the first * with */5. I will address your points as it seems that you haven't really thought about this. 1) In a production environment you should NOT be messing with the config. That's what test hardware is for. 2) The answer to this question is: crontab -e its really not that hard. I'm not running asterisk every minute. I'm looking to see if asterisk is running and then act accordingly 3) If asterisk fails believe me a full mailbox is the least of my worries. As for full logs I'd rather have more informationgrep awk are your friends. I prefer to keep things as simple as possible. Sure scripts like safe_asterisk are nice and do some really neat things but lets face it how often do you actually sit at the console of your asterisk box. My main PBX is located about 7 feet from my office desk and I still mostly use ssh (not even telnet) to get into the box. Mark C http://www.psh-inc.com Tzafrir Cohen wrote: On Fri, May 04, 2007 at 01:59:41PM -1000, Mark Coccimiglio wrote: What I do is add an entry in the crontab file as such: * * * * * if [ ! `/bin/pidof -s asterisk` ] ; then /usr/sbin/asterisk; fi Its simple and it works. Additionally if asterisk crashes then cron restarts the server in about a minute. Just be careful with your configs. It will not Just Work, because: 1. you may want to give Asterisk other command-line parameters (-p, -U) and not do that through asterisk.conf . 2. You may have your own reasons for wanting to stop asterisk occasionally. Having it run every minute from a cron job is a source for problems. 3. In case running asterisk generates an error, you get a very ugly flood in your logs and your mailbox. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR(accountcode) empty in * 1.4.4 (for local chan)
Steve,
I didnt mean to say that your patch did that. Actually i did saw this error
before applying your patch. i just mentioned it here. So is this problem
fixable?
On 5/5/07, Steve Murphy [EMAIL PROTECTED] wrote:
On Fri, 2007-05-04 at 15:25 +0500, Rizwan Hisham wrote:
Nops. removing res_features doesnt work.
Rizwan--
This is strange; It would seem your main/cdr.c and res/res_features.c
are out of sync!
The code chunk I sent you does not contain any references to
ast_cdr_merge, and
does not have anything to do with res_features... so... you should have
seen this
problem with or without my patch! Can you investigate and make sure
something hasn't been mixed into your release?
murf
On 5/4/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Nops. Its not working. i have restored to original chan_local
file. Im also having another problem now (in asterisk 1.4.4).
The call originates fine, ringing is done, call is accepted,
channels bridged fine. but when either of the channels hangup,
asterisk dies and displays the following msg:
asterisk: symbol lookup
error: /usr/lib/asterisk/modules/res_features.so: undefined
symbol: ast_cdr_merge
again, i dont know whats the problem. i'll try n remove the
res_features and then try caling again. Can anybody tell me
what other things will be effected by removing the
res_features?
On 5/3/07, Steve Murphy [EMAIL PROTECTED] wrote:
On Thu, 2007-05-03 at 16:47 +0500, Rizwan Hisham
wrote:
Hi all,
i just updated to asterisk 1.4.4 from 1.4.2. i was
doing this to
forward an unanswered call in 1.4.2
exten= 1,1,Dial(SIP/123,,Ttg)
exten= 1,2,Gotoif($[${DIALSTATUS}=ANSWERED]?:10)
exten= 1,3,Hangup
exten= 1,10,Dial(Local/2,,Ttg)
exten= 1,11,Hangup
exten= 2,1,Dial(SIP/234,,Ttg)
exten= 2,2,Hangup
All the CDR variables for the first channel
(SIP/123) are fine. but
when local channel initiates, it does not copy the
CDR(accountcode)
variable from the first channel (in asterisk 1.4.4),
whereas it did in
1.4.2. so the CDR(accountcode) variable for local
channel is empty in
1.4.4. This is a big problem for me as i have to
charge the forwarded
calls also and all calls are charged based on
account code. If
accountcode is empty, i cant make a decision how to
charge the call.
Can anybody fix this for me or do i have to jump
back to asterisk
1.4.2?
--
Regards
Rizwan Hisham
Software Engineer
Riswan--
This could easily be my fault. I've attached a fix,
that I can commit to
the source, if it works for you.
Here the instructions:
1. save the attachment to a file.
2. cd to your 1.4-source/channels directory
3. patch -p0 localfix
4. cd ..
5. make
6. make install
test
If there's no differences, you still have the same
problem, you'd best
restore the source to it's previous condition:
1. cd 1.4-sourcedir/channels
2. mv chan_local.c.orig chan_local.c
3. cd ..
4. make
5. make install
This patch will properly set the accountcode amaflag
from the local
channel's owner at channel creation time, and
therefore, the local
channels' CDR as well.
--
Steve Murphy
Software Developer
Digium
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--
Regards
Rizwan Hisham
Software Engineer
--
Regards
Rizwan Hisham
Software Engineer
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[asterisk-users] Queue Status
Hi I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TLS support
Hello, Does anybody know whether Asterisk 1.4 supports TLS? Or may be any work patches or branches? Thanks in advance -- Best Regards Alexander Olekhnovich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk x legacy pabx
Hi Josue, Yes you can use Asterisk along side an existing PABX. So with your existing Avaya you can allow it to connect to the handsets, but when calls are received for voicemail then you can send them to the Asterisk server another functionality you might find useful are conference rooms. Also if you place the Asterisk in front of the Avaya between it and the PSTN and Internet voip connection you can use Asterisk for IVR and for Least Cost Routing sending calls via Voip for cheaper connectivity. I hope this helps you. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti Sent: Saturday, 5 May 2007 1:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk x legacy pabx Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josué image001.gif___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager API Output
Hi, Is there any way that I can store my manager API output that is: My question is that is there any why using that I can get the QueueStatus and store the result in some text file for further processing. ?php $strHost = 127.0.0.1; $strUser = cron; $strSecret = 1234; $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die(Connection to host failed); fputs($oSocket, Action: Login\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: QueueStatus\r\n\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); fclose($oSocket); ? thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP registration problem
I've reposted with a more meaningful subject - hopefully someone will replyWe have an Asterisk v1.2.16 box registering with an ITSP using SIP. The registration succeeds, and is confirmed with SIP SHOW REGISTER. However, we frequently (every few minutes) see this on our console: REGISTER attempt 1 to [EMAIL PROTECTED] REGISTER attempt 2 to [EMAIL PROTECTED] Any ideas what is going on? In particular 1. What causes the two register attempt messages above? 2. Why is our asterisk box being associated with the entryunauthorized context, not the entryinternal context? (See below) 3. Why is the contact sip:[EMAIL PROTECTED]:5060 in our SIP messages, why s@ anything? Thanks MD -- Contents of sip.conf at ITSP: [999] context=entryinternal ; I know this context exists! This is the right context. type=friend username=999 secret= callerid=Test 999 host=dynamic nat=no canreinvite=no allow=ulaw allow=alaw dtmfmode=rfc2833 --- Console log from ITSP show strange SIP traffic: --- Scheduling destruction of call mailto:'[EMAIL PROTECTED]' '[EMAIL PROTECTED]' in 15000 ms pbx*CLI pbx*CLI -- SIP read from 123.183.86.231:5060: REGISTER sip:pbx.itsp.com SIP/2.0 Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;rport From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED] Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=999, realm=pbx.itsp.com, algorithm=MD5, uri=sip:pbx.itsp.com, nonce=5cec66c0, response=6451967016fc38f896efeb7247523fe1, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060 Event: registration Content-Length: 0 --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 123.183.86.231 : 5060 (NAT) Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=506 0 From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED] Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=506 0 From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED];tag=as7d680d48 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060;expires=120 Date: Fri, 04 May 2007 19:27:58 GMT ontent-Length: 0 -- SIP read from 123.183.86.231:5060: OPTIONS sip:pbx.itsp.com SIP/2.0 Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;rport From: asterisk sip:[EMAIL PROTECTED];tag=as6e5334cf To: sip:pbx.itsp.com Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 04 May 2007 19:38:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- (12 headers 0 lines) --- Looking for s in entryunauthorized (domain pbx.itsp.com) Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;received=123.183.86.231;rport=506 0 From: asterisk sip:[EMAIL PROTECTED];tag=as6e5334cf To: sip:pbx.itsp.com;tag=as51d476cd Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:74.110.57.25 Accept: application/sdp Content-Length: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue and voicemail
Hi all, thanks for this reply. Follows below the current configurations of mine asterisk, where the line functions perfectly, but does not obtain to rotear in case that no agent takes care of, for the voicemail. How it could give an option to the caller so that it can send a message? Sample: it dials 0 to send a message. Best Regards extensions.conf [0023] exten=s,1,answer exten=s,2,queue(35270023|Ttn|||32) exten=s,3,playback(working) exten=s,4,waitmusiconhold(10) exten=s,5,goto(0122,s,1) exten=s,6,goto(macro-voicemail,s,1) agents.conf agent = 1122,1122,Agent 001 agent = 1123,1123,Agent 002 agent = 1023,1023,Agent 003 agent = 1027,1027,Agent 004 agent = 0217,0217,Agent 005 queues.conf [general] [0023] music=default timeout=60 ;retry=1 leavewhenempty = yes maxlen = 6 joinempty = yes announce-frequency = 90 announce-holdtime = yes queue-minutes = queue-minutes strategy=roundrobin member = Agent/1023 member = Agent/1027 member = Agent/1122 member = Agent/1123 ;member = Agent/0217 2007/5/5, 0xception [EMAIL PROTECTED]: You could alternatively set a context for your queue in your config and create an extension for voicemail, if you would rather give the option to go to voice mail to the caller... (example: They can dial 0 to leave a message) On 5/4/07, Per Jessen [EMAIL PROTECTED] wrote: Josué Conti wrote: Hi all, good? I would like to know if the option exists to together integrate the function of queue with the voicemail of the agent, or the pilot of the group. For example, in case that none of the agents of queue obtains to take care of a call, this call would be directed for a voicemail. I think you can do this in the dialplan - set a timeout for the queue, then route to voicemail when it expires. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.comhttp://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API Output
Arun Kumar wrote: Is there any way that I can store my manager API output that is: My question is that is there any why using that I can get the QueueStatus and store the result in some text file for further processing. ?php $strHost = 127.0.0.1; $strUser = cron; $strSecret = 1234; $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die(Connection to host failed); fputs($oSocket, Action: Login\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: QueueStatus\r\n\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); fclose($oSocket); ? Is this a PHP question? You need to use fread() to get the output from Asterisk. Have a look at http://us.php.net/manual/en/function.fsockopen.php There are many examples. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk x legacy pabx
Hi Dean, thank you will be this attention. Currently asterisk is interconnected in pabx legacy through a A104D with protocol ISDN Qsig, uses LCR for routes of lesser cost and calls for other localities. But I see that domains of asterisk is limitless therefore would like to use it as also voicemail server, but as it would obtain to program the easiness? Best Regards Josué 2007/5/5, Dean Collins [EMAIL PROTECTED]: Hi Josue, Yes you can use Asterisk along side an existing PABX. So with your existing Avaya you can allow it to connect to the handsets, but when calls are received for voicemail then you can send them to the Asterisk server another functionality you might find useful are conference rooms. Also if you place the Asterisk in front of the Avaya between it and the PSTN and Internet voip connection you can use Asterisk for IVR and for Least Cost Routing sending calls via Voip for cheaper connectivity. I hope this helps you. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph [image: Call Button]http://click.mexuar.com/webuser/click/7/userurl/Cognationhttp://click.mexuar.com/webuser/nojs/7/userurl/Cognation -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Josué Conti *Sent:* Saturday, 5 May 2007 1:39 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Asterisk x legacy pabx Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josué ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users inline: image001.gif___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API Output
On Sat, 5 May 2007, Arun Kumar wrote: Hi, Is there any way that I can store my manager API output that is: Read The Fine WiKi!!! http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+PHP Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk x legacy pabx
Ive done a lot of work with Avaya. Voicemail systems attaché dot Avaya use Qsig trunk to pass calls to voicemail servers. The core of their modular messaging/message networking infrastructure can also use VPIM for communication between vmail servers. As far as I know, you cant use Asterisk in the same way you can use a modular messaging setup. Asterisk will only work if you actually terminate the employees phone on the asterisk box and that would be kind of pointless because businesses only want Avaya because eof the extra feature they offer. You would of course lose most of them if you were just using Avaya to manage the tie lines to an Asterisk box. On the brighter side, I would bet your licensing would be a hell of a lot cheaper I worked with Avaya for 3 years prototyping solutions involving their CCS/SES product line. Their stuff does not play well with other equipment. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti Sent: Saturday, May 05, 2007 1:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk x legacy pabx Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josué ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P usada?
Alguien tiene una TDM400P con modulo FXS usada a la venta ??, obviamente a precio de tarjeta usada... saludos, Rodrigo Mercado S. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04
Mark Coccimiglio wrote: Tzafrir; Actually I have found this config to work really well. I prefer to use a script run from inittab but Ubuntu doesn't work like Redhat or BSD. On a production box keeping asterisk up and running is THE TOP priority. If you would rather check every five minutes then replace the first * with */5. I will address your points as it seems that you haven't really thought about this. 1) In a production environment you should NOT be messing with the config. That's what test hardware is for. 2) The answer to this question is: crontab -e its really not that hard. I'm not running asterisk every minute. I'm looking to see if asterisk is running and then act accordingly 3) If asterisk fails believe me a full mailbox is the least of my worries. As for full logs I'd rather have more informationgrep awk are your friends. I prefer to keep things as simple as possible. Sure scripts like safe_asterisk are nice and do some really neat things but lets face it how often do you actually sit at the console of your asterisk box. My main PBX is located about 7 feet from my office desk and I still mostly use ssh (not even telnet) to get into the box. at least on ubuntu 6.10 safe_asterisk requires one simple fix, not really a headbreaker (something with output redirection). You could actually edit the script to not start a console if you dont' want it to (say for security reasons). If you wanted to start asterisk and keep monitoring it, that is what init is for. I don't know about ubuntu startup, but traditional sysV init would simply restart a process if it ever quits (respawn). My bet is that startup can do the same somehow, this is a far better way to keep * up -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Virtual IP Adresses and SIP requests failing...
Hello - Well I've been able to find a bit more about my problem. Again - I am not bound to a specific interface (0.0.0.0) When a SIP invite addressed to the .36 address, Asterisk replies FROM the .38 address. Is this the expected behavior? Wouldn't it make sense for Asterisk to reply on FROM the IP it rec'd on? Any thoughts??? Is this a bug? On 5/3/07, Christopher Aloi [EMAIL PROTECTED] wrote: Hey All: Question; when using a virtual IP on an Asterisk server, I am having trouble getting sip user to register to the ViP. They are able to register with the true IP, just not the virtual. It seems Asterisk is rejecting the SIP invite, register, etc (like it's not destined for this server) I've added all the IP's to the domain listing in sip.conf and in the Asterisk console a sip show domains shows both the virtual and the physical IP. Am I missing something? I have Asterisk bound to 0.0.0.0 which should tell it to listen on all IP's, right?? Some Details: ## ifconfig eth1 - inet addr:69.67.250.38 eth1:0 - inet addr: 69.67.250.36 (ViP) ## sip.conf [general] domain=69.67.250.36 domain=69.67.250.38 bindport=5060 port=5060 bindaddr=0.0.0.0 ## sip show domains Our local SIP domains: Context Set by 69.67.250.36 (default) [Configured] 69.67.250.38 (default) [Configured] ## tshark -i eth1 -R sip ## Call to .38 10.818719 66.218.1.47 - 69.67.250.38 SIP Request: OPTIONS sip: 69.67.250.38 10.818903 69.67.250.38 - 66.218.1.47 SIP Status: 200 OK 10.820676 192.168.0.102 - 69.67.250.38 SIP Request: OPTIONS sip: 69.67.250.38 10.821626 69.67.250.38 - 192.168.0.102 SIP Status: 200 OK 10.829019 66.218.1.47 - 69.67.250.38 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 10.830792 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 10.835473 66.218.1.47 - 69.67.250.38 SIP Request: ACK sip:[EMAIL PROTECTED] 10.841651 66.218.1.47 - 69.67.250.38 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] , with session description 10.841880 69.67.250.38 - 66.218.1.47 SIP Status: 100 Trying 10.847744 69.67.250.38 - 69.67.248.83 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 10.847874 69.67.250.38 - 66.218.1.47 SIP/SDP Status: 183 Session Progress, with session description 10.848852 69.67.248.83 - 69.67.250.38 SIP Status: 100 Trying 16.724167 69.67.248.83 - 69.67.250.38 SIP/SDP Status: 183 Session Progress, with session description 16.725928 69.67.248.83 - 69.67.250.38 SIP/SDP Status: 200 OK, with session description 16.726053 69.67.250.38 - 69.67.248.83 SIP Request: ACK sip:[EMAIL PROTECTED]:5060 16.726373 69.67.250.38 - 66.218.1.47 SIP/SDP Status: 200 OK, with session description 16.731913 66.218.1.47 - 69.67.250.38 SIP Request: ACK sip:[EMAIL PROTECTED] 19.561514 69.67.248.83 - 69.67.250.38 SIP Request: BYE sip:[EMAIL PROTECTED] 19.561617 69.67.250.38 - 69.67.248.83 SIP Status: 200 OK 19.562158 69.67.250.38 - 66.218.1.47 SIP Request: BYE sip:[EMAIL PROTECTED] :5004;transport=udp 19.565798 66.218.1.47 - 69.67.250.38 SIP Status: 200 OK ## Call to .36 90.821676 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 90.821873 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 91.321664 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 91.822061 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 92.322452 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 92.821931 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 94.323765 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 94.452850 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 94.453240 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 94.822695 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 98.324204 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 98.453399 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 98.822235 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 102.325048 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 102.821775 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 106.325130 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 106.822293 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 110.326101 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 110.587025 66.218.1.47 - 69.67.250.36 SIP Request: CANCEL sip:[EMAIL PROTECTED] 110.587101 69.67.250.38 -
Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04
On Sat, May 05, 2007 at 06:23:43PM +0200, Remco Post wrote: Mark Coccimiglio wrote: Tzafrir; Actually I have found this config to work really well. I prefer to use a script run from inittab but Ubuntu doesn't work like Redhat or BSD. On a production box keeping asterisk up and running is THE TOP priority. If you would rather check every five minutes then replace the first * with */5. I will address your points as it seems that you haven't really thought about this. 1) In a production environment you should NOT be messing with the config. That's what test hardware is for. 2) The answer to this question is: crontab -e its really not that hard. I'm not running asterisk every minute. I'm looking to see if asterisk is running and then act accordingly 3) If asterisk fails believe me a full mailbox is the least of my worries. As for full logs I'd rather have more informationgrep awk are your friends. I prefer to keep things as simple as possible. Sure scripts like safe_asterisk are nice and do some really neat things but lets face it how often do you actually sit at the console of your asterisk box. My main PBX is located about 7 feet from my office desk and I still mostly use ssh (not even telnet) to get into the box. at least on ubuntu 6.10 safe_asterisk requires one simple fix, not really a headbreaker (something with output redirection). Bashism? The rule in Debian is that a bourne shell script (#!/bin/sh) should not use bash-specific features, such as . If it does, it should explicitly ask for bash: '#!/bin/bash' You could actually edit the script to not start a console if you dont' want it to (say for security reasons). Could you please elaborate? I believe that this would wreck the error handling in that script. If you wanted to start asterisk and keep monitoring it, that is what init is for. I don't know about ubuntu startup, but traditional sysV init would simply restart a process if it ever quits (respawn). My bet is that startup can do the same somehow, this is a far better way to keep * up But this means editing /etc/inittab every time you actually want to stop asterisk. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P usada?
On May 5, 2007, at 12:06 PM, Rodrigo Mercado wrote: Alguien tiene una TDM400P con modulo FXS usada a la venta ??, obviamente a precio de tarjeta usada... saludos, Rodrigo Mercado S. For anyone who is not a Spanish speaker, Rodrigo is looking for a used TDM400P card with FXS modules. He is expecting a price that would correspond with a used card. (In other words, cheap) Rodrigo: 1.) ¿Donde estás? ¿Cómo podria alguien dar un precio sin saber donde tendria que mandarlo? ¿España? ¿Puerto Rico? ¿Argentina? 2.) Si no hablas Inglés, seria mejor buscar una lista de Asterisk en Español, porque la mayoria de las personas aqui no hablen Español. Tom___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk telemarketer torture sound files
Hi, I have some annoying telemarketer calling me on a recurring basis, but I'd like to discourage them a bit and have some fun. I found this: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture but the link to download the sound files is dead (wyoming.e-tools.com is NXDOMAIN). Anyone have a copy of these? -Adam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk telemarketer torture sound files
Adam Jacob Muller wrote on 5/5/07 1:06 PM: Hi, I have some annoying telemarketer calling me on a recurring basis, but I'd like to discourage them a bit and have some fun. I found this: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture but the link to download the sound files is dead (wyoming.e-tools.com is NXDOMAIN). Anyone have a copy of these? I believe they're included in Asterisk's extra sounds package now. Look for the sounds with a tt- prefix on the filenames. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue and voicemail
You can add another line like exten=0,1,VoiceMail([EMAIL PROTECTED]) this will catch the dialing of 0 before or after it enters the queue... but if you want them to be able to do that while in the queue then you need to add to your queue config a line like context=your-queue-context and then create a context that has that extension like above in it. that way when someone dials 0 in the queue they are also redirected to the voicemail ... hopefully that helps you out... i could show you what i did for my queue setup but it's all in AEL so not sure if that would help you as much... but the idea is still the same On 5/5/07, Josué Conti [EMAIL PROTECTED] wrote: Hi all, thanks for this reply. Follows below the current configurations of mine asterisk, where the line functions perfectly, but does not obtain to rotear in case that no agent takes care of, for the voicemail. How it could give an option to the caller so that it can send a message? Sample: it dials 0 to send a message. Best Regards extensions.conf [0023] exten=s,1,answer exten=s,2,queue(35270023|Ttn|||32) exten=s,3,playback(working) exten=s,4,waitmusiconhold(10) exten=s,5,goto(0122,s,1) exten=s,6,goto(macro-voicemail,s,1) agents.conf agent = 1122,1122,Agent 001 agent = 1123,1123,Agent 002 agent = 1023,1023,Agent 003 agent = 1027,1027,Agent 004 agent = 0217,0217,Agent 005 queues.conf [general] [0023] music=default timeout=60 ;retry=1 leavewhenempty = yes maxlen = 6 joinempty = yes announce-frequency = 90 announce-holdtime = yes queue-minutes = queue-minutes strategy=roundrobin member = Agent/1023 member = Agent/1027 member = Agent/1122 member = Agent/1123 ;member = Agent/0217 2007/5/5, 0xception [EMAIL PROTECTED]: You could alternatively set a context for your queue in your config and create an extension for voicemail, if you would rather give the option to go to voice mail to the caller... (example: They can dial 0 to leave a message) On 5/4/07, Per Jessen [EMAIL PROTECTED] wrote: Josué Conti wrote: Hi all, good? I would like to know if the option exists to together integrate the function of queue with the voicemail of the agent, or the pilot of the group. For example, in case that none of the agents of queue obtains to take care of a call, this call would be directed for a voicemail. I think you can do this in the dialplan - set a timeout for the queue, then route to voicemail when it expires. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.comhttp://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P usada?
Chile. No hay listas en español, y si lo enviè en español es justamente porque si alguien no lo habla no puede estar en CHILE, de todas formas muchas gracias por la amabilidad de traducir mi correo. saludos, bye bye On 5/5/07, Tom Rymes [EMAIL PROTECTED] wrote: On May 5, 2007, at 12:06 PM, Rodrigo Mercado wrote: Alguien tiene una TDM400P con modulo FXS usada a la venta ??, obviamente a precio de tarjeta usada... saludos, Rodrigo Mercado S. For anyone who is not a Spanish speaker, Rodrigo is looking for a used TDM400P card with FXS modules. He is expecting a price that would correspond with a used card. (In other words, cheap) Rodrigo: 1.) ¿Donde estás? ¿Cómo podria alguien dar un precio sin saber donde tendria que mandarlo? ¿España? ¿Puerto Rico? ¿Argentina? 2.) Si no hablas Inglés, seria mejor buscar una lista de Asterisk en Español, porque la mayoria de las personas aqui no hablen Español. Tom___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk telemarketer torture sound files
On May 5, 2007, at 1:15 PM, Dave Miller wrote: Adam Jacob Muller wrote on 5/5/07 1:06 PM: Hi, I have some annoying telemarketer calling me on a recurring basis, but I'd like to discourage them a bit and have some fun. I found this: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture but the link to download the sound files is dead (wyoming.e- tools.com is NXDOMAIN). Anyone have a copy of these? I believe they're included in Asterisk's extra sounds package now. Look for the sounds with a tt- prefix on the filenames. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ Unfortunately, this doesn't seem to be the case :/ -=[~/asterisk-extra-sounds-en-gsm-current]=- -=[Sat May 05]=- -= [13:32:42]=- [EMAIL PROTECTED] ls -l tt-* ls: tt-*: No such file or directory checked through the files as well, i don't see them here or in the core sounds, though there are a few tt-* files in the core package tt-allbusy.gsm tt-monkeys.gsm tt-monkeysintro.gsm tt-somethingwrong.gsm tt-weasels.gsm -Adam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04
Tzafrir Cohen wrote: On Sat, May 05, 2007 at 06:23:43PM +0200, Remco Post wrote: Mark Coccimiglio wrote: Tzafrir; Actually I have found this config to work really well. I prefer to use a script run from inittab but Ubuntu doesn't work like Redhat or BSD. On a production box keeping asterisk up and running is THE TOP priority. If you would rather check every five minutes then replace the first * with */5. I will address your points as it seems that you haven't really thought about this. 1) In a production environment you should NOT be messing with the config. That's what test hardware is for. 2) The answer to this question is: crontab -e its really not that hard. I'm not running asterisk every minute. I'm looking to see if asterisk is running and then act accordingly 3) If asterisk fails believe me a full mailbox is the least of my worries. As for full logs I'd rather have more informationgrep awk are your friends. I prefer to keep things as simple as possible. Sure scripts like safe_asterisk are nice and do some really neat things but lets face it how often do you actually sit at the console of your asterisk box. My main PBX is located about 7 feet from my office desk and I still mostly use ssh (not even telnet) to get into the box. at least on ubuntu 6.10 safe_asterisk requires one simple fix, not really a headbreaker (something with output redirection). Bashism? The rule in Debian is that a bourne shell script (#!/bin/sh) should not use bash-specific features, such as . If it does, it should explicitly ask for bash: '#!/bin/bash' hmmm, you might have a point there, never thought of that. You could actually edit the script to not start a console if you dont' want it to (say for security reasons). Could you please elaborate? Change: CONSOLE=yes # Whether or not you want a console To 'CONSOLE=no' I believe that this would wreck the error handling in that script. If you wanted to start asterisk and keep monitoring it, that is what init is for. I don't know about ubuntu startup, but traditional sysV init would simply restart a process if it ever quits (respawn). My bet is that startup can do the same somehow, this is a far better way to keep * up But this means editing /etc/inittab every time you actually want to stop asterisk. Or change runlevel... well that is maybe a bit to much AIX :) -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SLA broken in 1.4.3?
SLA requires meetme which requires at a minimum ztdummy. So, you must compile and install zaptel, then compile and install asterisk 1.4.3 and the sla commands will be in the CLI. Let me know if you need help setting up SLA on Polycom phones with *. I've done it successfully and have the configs. I'm going to put them on the web some time soon. -Original Message- From: Jay Austad [mailto:[EMAIL PROTECTED] Sent: Friday, May 04, 2007 8:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SLA broken in 1.4.3? I configured my sla.conf to use with a Polycom phone. I have no idea if I did it right, however, none of the console sla commands exist. Do I have to something special to compile in this support, or should it just work out of the box? ~jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GXV-3000 IP Video Phone
Hello All, I just received some test units of Grandstream GXV-3000 IP Video Phone. I did some research and looks like Asterisk 1.2 does not support video H.264 but Asterisk 1.4 does. Is it correct? Actually I did try to test with Asterisk 1.2 and video did not initialize but voice worked... Any advice? Thanks, Nitesh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk x legacy pabx
Hi Salvatore, thanks for reply. And if pabx legacy was Siemens model HiPath 3750, could use asterisk as serving of voicemail and other applications? Best Regards Josué 2007/5/5, Salvatore Giudice [EMAIL PROTECTED]: I've done a lot of work with Avaya. Voicemail systems attaché dot Avaya use Qsig trunk to pass calls to voicemail servers. The core of their modular messaging/message networking infrastructure can also use VPIM for communication between vmail servers. As far as I know, you can't use Asterisk in the same way you can use a modular messaging setup. Asterisk will only work if you actually terminate the employee's phone on the asterisk box and that would be kind of pointless because businesses only want Avaya because eof the extra feature they offer. You would of course lose most of them if you were just using Avaya to manage the tie lines to an Asterisk box. On the brighter side, I would bet your licensing would be a hell of a lot cheaper… I worked with Avaya for 3 years prototyping solutions involving their CCS/SES product line. Their stuff does not play well with other equipment. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com http://voipsecuritytraining.com/ 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Josué Conti *Sent:* Saturday, May 05, 2007 1:39 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Asterisk x legacy pabx Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josué ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!
Thanks I did that as well. I did however get the problem fixed by setting canreinvote=yes Apparently the polycom wants it when the soft phones don't Sorry, I meant canreinvite can re inVOTE- is something that dead people do in my home state of Mississippi. G -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brandon Kruse Sent: Thursday, May 03, 2007 4:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk-Polycom HEPPP Do not use the transfer key on the Polycom. Use /etc/asterisk/features.conf and setup blind and attended transfers for asterisk. It just works better in my opinion. -bk - Original Message - From: Jim Suber [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, May 3, 2007 9:53:58 AM (GMT-0800) America/Tijuana Subject: [asterisk-users] Asterisk-PolycomHEPPP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RXFAX/TXFAX
But i want to do with a TDM400 or winh E1, using rxfax app (or something like this) from the dialplan, without hylafax or esoteric codes. My question is becouse i read than 1.4 supports T.38, and then should receive fax i guess... There is not something like app_rxfax / app_txfax ??? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Per Jessen Enviado el: sábado, 05 de mayo de 2007 2:30 Para: asterisk-users@lists.digium.com Asunto: Re: [asterisk-users] RXFAX/TXFAX Cesar Benjamin Garcia Martinez wrote: Somebody can tell me, what way i can send/receive faxes with asterisk 1.4??? [snip] How to i can send/receive fax to/from PSTN on asterisk 1.4 ? Check out a very recent thread on just that subject. Or go study how to use iaxmodem and hylafax. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Informacin de NOD32, revisin 2243 (20070505) __ Este mensaje ha sido analizado con NOD32 antivirus system http://www.nod32.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM400P usada?
Bueno, no esta en chile, yo no estoy en chile, pero hablo español, soy de mexico, asi que en parte tienes razón, pero tb creo que deberías haber puesto de donde eres. De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Rodrigo Mercado Enviado el: sábado, 05 de mayo de 2007 12:38 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] TDM400P usada? Chile. No hay listas en español, y si lo enviè en español es justamente porque si alguien no lo habla no puede estar en CHILE, de todas formas muchas gracias por la amabilidad de traducir mi correo. saludos, bye bye On 5/5/07, Tom Rymes [EMAIL PROTECTED] wrote: On May 5, 2007, at 12:06 PM, Rodrigo Mercado wrote: Alguien tiene una TDM400P con modulo FXS usada a la venta ??, obviamente a precio de tarjeta usada... saludos, Rodrigo Mercado S. For anyone who is not a Spanish speaker, Rodrigo is looking for a used TDM400P card with FXS modules. He is expecting a price that would correspond with a used card. (In other words, cheap) Rodrigo: 1.) ¿Donde estás? ¿Cómo podria alguien dar un precio sin saber donde tendria que mandarlo? ¿España? ¿Puerto Rico? ¿Argentina? 2.) Si no hablas Inglés, seria mejor buscar una lista de Asterisk en Español, porque la mayoria de las personas aqui no hablen Español. Tom___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNow!
Ed Nuñez wrote: Does anyone know how to gain access directly to the configuration files in AsteriskNow? I have dual NICs and need to change the binding in the config file. I believe they blocked ssh2 access by default. ssh is not blocked. You have to ssh into the userid admin. If you haven't changed it the password is password. To get root access type sudo su. Once there you can change the root password with passwd. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RXFAX/TXFAX
On Sat, May 05, 2007 at 02:25:50PM -0500, Cesar Benjamin Garcia Martinez wrote: My question is becouse i read than 1.4 supports T.38, and then should receive fax i guess... 1.4 only supports VoIP passthrough of T.38 . That is: if you get a T.38 fax, it can be safely redirected to a T.38-capable device. But Asterisk cannot handle that fax by itself. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?
At the very least, he's abusing his customers. Substances? I hadn't thought of that. On 4/30/07, Salvatore Giudice [EMAIL PROTECTED] wrote: I suspect that Jed has a substance abuse problem and that he may be in rehab. I don't know for sure of course. This kind of silence is indicative of people being hauled back to rehab. Anyway, maybe he just makes a habit of running off with people's money. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Monday, April 30, 2007 2:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip? On 2007-03-26 01:46:40 -0700, Salvatore Giudice [EMAIL PROTECTED] said: This is a multi-part message in MIME format. I opened up a ticket with them, but I'm not holding my breath. I think it's time to start moving my DID's before the inbound stops working. That seems like it was probably wise and I hope you followed through. I am now unable (for a week or so) to dial any outbound calls, or receive any at my did. Additionally when trying to call them at there local phone I get the disconnected message. They provided by FAR the best call quality for me when they where working, so I am going to miss them if they are gone forever. Also, I still have like 24$ (us) credit with them... I still hope they return, but wouldn't count on it. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.4 and Custom Postgres 8.2.4 (checking for PQexec in -lpq... no)
Dear All, Why does my configure fail like so: checking for pg_config... /usr/local/pgsql/8.2.4/bin/pg_config checking for PQexec in -lpq... no configure: *** configure: *** The PostgreSQL installation on this system appears to be broken. configure: *** Either correct the installation, or run configure configure: *** including --without-postgres Configure options are: env CC=/usr/local/bin/gcc ./configure --with-ssl=/usr/local/ssl --with-postgres=/usr/local/pgsql/8.2.4 configure has found pg_config, what more does it need? I even tried: env CC=/usr/local/bin/gcc CPPFLAGS=-I/usr/local/pgsql/8.2.4/include \ LDFLAGS=-L/usr/local/pgsql/8.2.4/lib \ LD_LIBRARY_PATH=/usr/local/pgsql/8.2.4/lib ./configure --with-ssl=/usr/local/ssl --with-postgres=/usr/local/pgsql/8.2.4 Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNow!
On Sat, 5 May 2007, Bill Merriam wrote: To get root access type sudo su. Once there you can change the root password with passwd. Seems a bit redundant: -fs::sedwards:~$ man sudo NAME sudo - execute a command as another user -fs::sedwards:~$ man su NAME su - run a shell with substitute user and group IDs How about just sudo bash? (Yes, it is 2 characters longer, but you're only executing 2 programs instead of 3.) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I'm looking for solution
HI I have 3 Linksys SIP901 IP phones I also have a pc I'm not using it amd athlon 1800+ 512mb ram and 40 gb hdd I'm looking to connect this phones together and to make calls between them Not from outside of my lan I don't know how to configure asterisknow beta Can somebody help I'm doing this in my house to connect rooms With respect Ardit Saliu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk telemarketer torture sound files
Adam Jacob Muller wrote on 5/5/07 1:38 PM: On May 5, 2007, at 1:15 PM, Dave Miller wrote: Adam Jacob Muller wrote on 5/5/07 1:06 PM: Hi, I have some annoying telemarketer calling me on a recurring basis, but I'd like to discourage them a bit and have some fun. I found this: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture but the link to download the sound files is dead (wyoming.e-tools.com is NXDOMAIN). Anyone have a copy of these? I believe they're included in Asterisk's extra sounds package now. Look for the sounds with a tt- prefix on the filenames. Unfortunately, this doesn't seem to be the case :/ -=[~/asterisk-extra-sounds-en-gsm-current]=- -=[Sat May 05]=- -=[13:32:42]=- [EMAIL PROTECTED] ls -l tt-* ls: tt-*: No such file or directory checked through the files as well, i don't see them here or in the core sounds, though there are a few tt-* files in the core package tt-allbusy.gsm tt-monkeys.gsm tt-monkeysintro.gsm tt-somethingwrong.gsm tt-weasels.gsm Ah, those are the ones I was thinking of. For some reason I didn't think those would be in core :) -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I'm looking for solution
Ardit Saliu wrote: HI I have 3 Linksys SIP901 IP phones I also have a pc I’m not using it amd athlon 1800+ 512mb ram and 40 gb hdd I’m looking to connect this phones together and to make calls between them Not from outside of my lan I don’t know how to configure asterisknow beta Can somebody help I’m doing this in my house to connect rooms Have you looked at http://www.asterisknow.org/files/downloads/quickstart_asterisknow.pdf ? With respect Ardit Saliu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I'm looking for solution
HI I have 3 Linksys SIP901 IP phones I also have a pc I'm not using it amd athlon 1800+ 512mb ram and 40 gb hdd I'm looking to connect this phones together and to make calls between them Not from outside of my lan I don't know how to configure asterisknow beta Can somebody help I'm doing this in my house to connect rooms With respect Ardit Saliu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_config_pgsql.c in * 1.4.4
Dear All, Where can I find a res_pgsql.conf and some sql to insert for tables etc.? Are all db res things to be done via odbc now? Why was this included with no docs or sample conf? Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P usada?
Hey Look http://www.asterisk-es.org Best Regards On 5/5/07, Cesar Benjamin Garcia Martinez [EMAIL PROTECTED] wrote: Bueno, no esta en chile, yo no estoy en chile, pero hablo español, soy de mexico, asi que en parte tienes razón, pero tb creo que deberías haber puesto de donde eres. *De:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *En nombre de *Rodrigo Mercado *Enviado el:* sábado, 05 de mayo de 2007 12:38 *Para:* Asterisk Users Mailing List - Non-Commercial Discussion *Asunto:* Re: [asterisk-users] TDM400P usada? Chile. No hay listas en español, y si lo enviè en español es justamente porque si alguien no lo habla no puede estar en CHILE, de todas formas muchas gracias por la amabilidad de traducir mi correo. saludos, bye bye On 5/5/07, *Tom Rymes* [EMAIL PROTECTED] wrote: On May 5, 2007, at 12:06 PM, Rodrigo Mercado wrote: Alguien tiene una TDM400P con modulo FXS usada a la venta ??, obviamente a precio de tarjeta usada... saludos, Rodrigo Mercado S. For anyone who is not a Spanish speaker, Rodrigo is looking for a used TDM400P card with FXS modules. He is expecting a price that would correspond with a used card. (In other words, cheap) Rodrigo: 1.) ¿Donde estás? ¿Cómo podria alguien dar un precio sin saber donde tendria que mandarlo? ¿España? ¿Puerto Rico? ¿Argentina? 2.) Si no hablas Inglés, seria mejor buscar una lista de Asterisk en Español, porque la mayoria de las personas aqui no hablen Español. Tom___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ${ANSWEREDTIME} Broken on 1.2.13?
No matter what I do, ${ANSWEREDTIME} is always 0, even on the most
simplest dial plan such as:
Using Asterisk 1.2.13
exten = 77,1,Answer
exten = 77,2,Playback(custom/dax/S300) ; one minute file
exten = 77,3,Noop(${ANSWEREDTIME})
exten = 77,4,Hangup
-- Executing Answer(SIP/5402-b7b45f58, ) in new stack
-- Executing Playback(SIP/5402-b7b45f58, custom/dax/S300) in new
stack
-- Playing 'custom/dax/S300' (language 'en')
-- Executing NoOp(SIP/5402-b7b45f58, ) in new stack
-- Executing Hangup(SIP/5402-b7b45f58, ) in new stack
What gives on this simple thing?
Bart
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RE: [asterisk-users] OT: USB T1/E1 Interface?
To everybody: Thanks for your thoughts and suggestions. This will be my last post to this list on this subject. I've started a blog about my research into this project: http://myossjourneys.blogspot.com/ If you want to discuss this any further please do so over there. Thanks again! -MC -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, May 03, 2007 1:39 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: USB T1/E1 Interface? Way cool product. Way too cool for my neighborhood -- the interface box is $7k. Software will set you back $3k to $30k. And then I would have no clue what to do with it. Maybe we could interest the guy thats building his own open telco hardware: http://www.rowetel.com/ucasterisk/pr1.html He seems to have the skills :) On Thu, 3 May 2007, Jorge Mendoza wrote: http://www.gl.com/laptopt1.html Jorge Michael Collins wrote: Why? There used to be a saying 'usb is for mice, firewire is for men', though USB has grown a bit in bandwidth since then, it is still not very well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I suspect a lack of demand. Havng a E1 termintae in your laptop is quite useless, and a server usually has plenty of slots (if not, buy a bigger server ;-). Just for fun. I'm a telecom geek and having a USB T1 interface would be a fun toy to tinker with. Besides, it might lead to some useful products. -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial Plan for Multi-Location Support Queue
Hi, I am in the process of planning a dial plan, In regards to the requirement, I am confused how to go about the dial plan. The scenario is like below. BRANCH - A - (COMPANY) Line 1 -- Extension 239 Line 2 -- Extension 8239 BRANCH - B - (COMPANY) Line 1 -- Extension 239 Line 2 -- Extension 8239 Now what I need is that if a user in Branch - A wants to dial Branch - B, he just needs to use 88xxx(extension of Branch - B) Similarly, if a user in Branch - B wants to dial Branch - A, he just needs to use 89xxx(extension of Branch - A) In this regards, I am not sure how do I achieve inter brach connection using asterisk to fit my 88 89 prefix dial plan for multi-location. More over, said that, we will have a support Queue in Branch - A(extension 700), users from Branch - B should be able to join the Queue(extension 700) to accept support calls vice-versa, I dont know how this is possible what would my dial plans be. It would be much appreciated if someone can help me resolve this dial plan support issue. Thannks, Deepak - Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your freeaccount today.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channel / Exten Status
Hello guys, I am using the API Mananger with PHP to initiate a call from a webpage. First I call to a line number, and then to an asterisk extension. I followed examples on using the API Mananger, without any problem, and working great. Now I have a problem. I can initiate the call. I can call to the line number, and then to the extension. But what I need now, is to keep track of the status of this comunication. I am a bit lost on how should I check the Channel status, and how can I verify when the extension is being used by this person who initiated the call from a webpage, since this extension could also be used by another person, from the webpage or internally from a IP telephone. Could anyone point me in the right direction? THank you, Pablo___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Playback() to play a random sound file
steve, thats Great... my C is old and ftw operated differently on sysV, solaris, sunos, ultrix, and osf... so I went back to bourne... couldn't work through the idiosyncracies of gnu autoconf, etc... although I have a many reasons to, I just couldn't get to production 'C' coding level... daveC Steve Edwards wrote: Steve Edwards wrote: On Tue, 1 May 2007, Jay Austad wrote: I've got a directory under /var/lib/asterisk/sounds which contains a bunch of sound files. I would like to call the Playback command to play the files, but I need it to select a file to play randomly. Is there any way to do this? I do this with an AGI. On Wed, 2 May 2007, dave cantera wrote: here is a way that I solved a similar problem... have a shell script that runs and indexes all the files in the directory into an ascii flat file with a format of filename 0001 directory/tt-weasels 0002 directory/tt-monkeys in your dialplan use the rand() to pick a number, pass it to the shell script as an arg[], then the shells script grep()'s and cut()'s the filename puts it in a db varaible, the dialplan picks it up and plays it... as you can see, I haven't done it yet :) but, in theory it works... you could skip the dialplan rand() and just use linux rand based on the minutes or seconds value for current time... you don't have to zero fill the index either, I seem to like nicely formated files, they are easier for humans to read. daveC Sounds like a lot of effort to avoid writing an AGI. If you have the skills to write the script described above, you have the skills to write an AGI -- you can write AGI's in shell scripts, btw. AGI's accept stuff from Asterisk on stdin and send stuff back to Asterisk on stdout -- very simple and elegant actually. Take your script and rewrite the reading arguments bits to read from stdin and change the write db bits to write to stdout (set a channel variable) and you have an AGI and a much cleaner dialplan. I write AGI's in C for speed and flexibility. No interpreter (bash, perl, php, etc.) to fire up, full access to anything you want to do. In C, I call ftw() (ftw - traverse (walk) a file tree). If I get more than 1 file, I choose one randomly. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RXFAX/TXFAX
Ast 1.4 will pass through T.38, but not terminate/originate T38. Be sure you understand the implications for your fax termination MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cesar Benjamin Garcia Martinez Sent: Saturday, May 05, 2007 3:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] RXFAX/TXFAX But i want to do with a TDM400 or winh E1, using rxfax app (or something like this) from the dialplan, without hylafax or esoteric codes. My question is becouse i read than 1.4 supports T.38, and then should receive fax i guess... There is not something like app_rxfax / app_txfax ??? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Per Jessen Enviado el: sábado, 05 de mayo de 2007 2:30 Para: asterisk-users@lists.digium.com Asunto: Re: [asterisk-users] RXFAX/TXFAX Cesar Benjamin Garcia Martinez wrote: Somebody can tell me, what way i can send/receive faxes with asterisk 1.4??? [snip] How to i can send/receive fax to/from PSTN on asterisk 1.4 ? Check out a very recent thread on just that subject. Or go study how to use iaxmodem and hylafax. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Informacin de NOD32, revisin 2243 (20070505) __ Este mensaje ha sido analizado con NOD32 antivirus system http://www.nod32.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] auto call out via drop file ERROR: 'OutgoingSpoolFailed'
has anyone run into this message? for some reason, which I can not determine, this script stop working and now gives this error. I googled 'outgoingspoolfailed' but not too much turned up... only questions, no answers... :( I am mv'ng a .call file to the ./outgoing directory. the call initiates then hangs up... and the reason 0, in the last line below, just doesn't help too much... what it was doing was calling and playing a message regardless of being answered (but that is another day's problem)... today the script and .call file initiate a call but hangs up whether answered or not in about 4+/- seconds... as you can see below, hangup is called immediately and the 'failed' extension is then executed... but why is it now failing? any thoughts? daveC pbv01*CLI -- Hungup 'Zap/4-1' -- Executing [EMAIL PROTECTED]:1] NoOp(OutgoingSpoolFailed, Call Failed) in new stack -- Executing [EMAIL PROTECTED]:2] Set(OutgoingSpoolFailed, CALL_ACK=failed) in new stack -- Executing [EMAIL PROTECTED]:3] AGI(OutgoingSpoolFailed, lax/track-laxcalls.sh|failed|failed) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/lax/track-laxcalls.sh -- AGI Script lax/track-laxcalls.sh completed, returning 0 -- Executing [EMAIL PROTECTED]:4] Wait(OutgoingSpoolFailed, 1) in new stack [May 3 00:59:08] NOTICE[8878]: pbx_spool.c:341 attempt_thread: Call failed to go through, *reason 0* -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GXV-3000 IP Video Phone
nitesh, you are correct. you need 1.4.x... daveC Nitesh Divecha wrote: Hello All, I just received some test units of Grandstream GXV-3000 IP Video Phone. I did some research and looks like Asterisk 1.2 does not support video H.264 but Asterisk 1.4 does. Is it correct? Actually I did try to test with Asterisk 1.2 and video did not initialize but voice worked... Any advice? Thanks, Nitesh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about more than one drop file
shawn, you can set an archive variable in the .call file to 'yes' and it will save it in ./outgoing_done... if there is now outbound line availible, the .call file is updated (appended to) as per the status... * will keep trying till it completes the calls or the number of retries is reached. then it will archive the .call file if archive=yes... if you drop a ton of files in the ./outgoing, it tries to make all the calls at 'almost' once. if you drop 20 .call files in there in about 2 seconds, all calls will initiate. if you have less than 20 outbound lines, the will all get stalled (for lack of a better word) and queue up until an outbound line is freed up... daveC shawn bright wrote: hello there all, if i have a script that writes drop files into /var/spool/asterisk/outgoing asterisk picks up the file and initiates the call just fine. however, what is supposed to happen if more than one gets dropped in there within like a second. Will it wait till the first is complete to initiate the second ? Do they dissapear ? thanks shawn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.6.2/785 - Release Date: 05/02/2007 02:16 PM -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASA-2007-013: IAX2 users can cause unauthorizeddata disclosure
Has 1.2.19 been released ? - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: undisclosed-recipients: Sent: Friday, May 04, 2007 12:20 PM Subject: [asterisk-users] ASA-2007-013: IAX2 users can cause unauthorizeddata disclosure Asterisk Project Security Advisory - ASA-2007-013 +--+ | Product| Asterisk | |--+---| | Summary| IAX2 users can cause unauthorized data disclosure | |--+---| | Nature of Advisory | Unauthorized information disclosure | |--+---| |Susceptibility| Remote authenticated sessions | |--+---| | Severity | Low | |--+---| |Exploits Known| No | |--+---| | Reported On | April 27, 2007 | |--+---| | Reported By | Tim Panton, Mexuar, [EMAIL PROTECTED] | | | | | | Birgit Arkesteijn, Westhawk, [EMAIL PROTECTED] | |--+---| | Posted On | May 4, 2007 | |--+---| | Last Updated On| May 4, 2007 | |--+---| | Advisory Contact | [EMAIL PROTECTED] | |--+---| | CVE Name | CVE-2007-2488 | +--+ +--+ | Description | From: Tim Panton [EMAIL PROTECTED] | | | | | | Date: 27 April 2007 08:02:36 BDT | | | | | | To: Kevin P. Fleming [EMAIL PROTECTED] | | | | | | Subject: Possible IAX2 vulnerability (Minor) | | | | | | | | | | | | We've stumbled on a bug in the way Asterisk's IAX2 handles text | | | | | | frames. | | | | | | I'm emailing you because it is a borderline security | | | vulnerability, | | | | | | and my | | | | | | friends in the security world tell me that I should notify the | | | | | | vendor privately | | | | | | first. If you feel it isn't a security issue, let me know and| | | I'll | | | | | | put it in mantis. | | | | | | | | | | | | chan_iax2 assumes that the content of a text frame is a null | | | | | | terminated | | | | | | string (C style), and when time comes to forward the string it | | | uses | | | | | | strlen | | | | | | to determine the message length. | | | | | | | | | | | | If you send a frame without a 0 byte in it, Asterisk forwards a | | | | | | frame that | | | | | | includes the sent data and some extra (presumably heap) data.| | | | | | | | | | | | If an attacker were lucky, the extra data could contain | | | something | | | | | | interesting. | | | | | | Or conceivably it could cause a segmentation violation. | +--+ +--+ | Resolution | Asterisk code has been modified to enforce null-termination of | || incoming text frames received by the IAX2 channel driver| || (chan_iax2). When text frames are received without | || null-termination, this may result in the
RE: [asterisk-users] Asterisk x legacy pabx
Its basically the same problem. Asterisk is not a standalone voicemail server. It would have to support Qsig. Asterisk doea not exactly have expansive Qsig support. I believe there are several bounties out for Qsig. Without Qsig, you would have to use parallel forking and ring the users avaya or siemans extension and also the same extension on an Asterisk box. Youd have to manage a dummy number for every mailbox configure dont he asterisk box. Also, I dont know for siemans, but Avaya doesnt support parallel forking, so you would have to either configure both the employee and asterisk as an optum extension or buy an x-mobility/extension-to-celluar license to accomplish either. I think x-mobility is $300 list per phone. Its horribly expensive. Among Nortel, Avaya, Mitel voicemail systems Mitel is by far the best product of these 3. Avaya message networking/modular messaging is basically a beta. Nobody should consider that GA. Its horrible. Nortel requires too much professional service money to get up and running. Nortel seems to think they can charge 4 times more for everything because it says Nortel. I havent figured that one out yet. Mitel was my preferred vendor voicemail product since it is reasonably priced and their support organization is actually attentive. Check out Mitel 10. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti Sent: Saturday, May 05, 2007 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk x legacy pabx Hi Salvatore, thanks for reply. And if pabx legacy was Siemens model HiPath 3750, could use asterisk as serving of voicemail and other applications? Best Regards Josué 2007/5/5, Salvatore Giudice [EMAIL PROTECTED]: I've done a lot of work with Avaya. Voicemail systems attaché dot Avaya use Qsig trunk to pass calls to voicemail servers. The core of their modular messaging/message networking infrastructure can also use VPIM for communication between vmail servers. As far as I know, you can't use Asterisk in the same way you can use a modular messaging setup. Asterisk will only work if you actually terminate the employee's phone on the asterisk box and that would be kind of pointless because businesses only want Avaya because eof the extra feature they offer. You would of course lose most of them if you were just using Avaya to manage the tie lines to an Asterisk box. On the brighter side, I would bet your licensing would be a hell of a lot cheaper I worked with Avaya for 3 years prototyping solutions involving their CCS/SES product line. Their stuff does not play well with other equipment. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com http://voipsecuritytraining.com/ 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti Sent: Saturday, May 05, 2007 1:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk x legacy pabx Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josué ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk telemarketer torture sound files
Just forward them to 1-800-big-dick or some other 800 toll free phone sex line. They can't tell they've been forwarded. They'll figure it out eventually. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Jacob Muller Sent: Saturday, May 05, 2007 1:07 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk telemarketer torture sound files Hi, I have some annoying telemarketer calling me on a recurring basis, but I'd like to discourage them a bit and have some fun. I found this: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture but the link to download the sound files is dead (wyoming.e-tools.com is NXDOMAIN). Anyone have a copy of these? -Adam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?
My money is on compulsory drug rehab or simply being held for 45 days of observation after being caught sexually abusing a pony. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Lynn Sent: Saturday, May 05, 2007 4:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip? At the very least, he's abusing his customers. Substances? I hadn't thought of that. On 4/30/07, Salvatore Giudice mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I suspect that Jed has a substance abuse problem and that he may be in rehab. I don't know for sure of course. This kind of silence is indicative of people being hauled back to rehab. Anyway, maybe he just makes a habit of running off with people's money. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Monday, April 30, 2007 2:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip? On 2007-03-26 01:46:40 -0700, Salvatore Giudice mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] said: This is a multi-part message in MIME format. I opened up a ticket with them, but I'm not holding my breath. I think it's time to start moving my DID's before the inbound stops working. That seems like it was probably wise and I hope you followed through. I am now unable (for a week or so) to dial any outbound calls, or receive any at my did. Additionally when trying to call them at there local phone I get the disconnected message. They provided by FAR the best call quality for me when they where working, so I am going to miss them if they are gone forever. Also, I still have like 24$ (us) credit with them... I still hope they return, but wouldn't count on it. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
