[asterisk-users] how to detect pickup...
Hello asterisk-users, My SIP phones are in pickupgroup, and if some of them ringing from other phone can pick up with *8 as usual. But I want to know if this happen. I've tried the a extension, but seems not working. Any other idea? -- Best regards, Gergo mailto:[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to detect pickup...
Hi, One of the solutions would be to overwrite standard *8 behaviour with your custom macro that will 1) pickup a call as usual b) send notification via AMI or whatever else you want. This can be done with [applicationmap] in features.conf - see http://www.voip-info.org/wiki-Asterisk+config+features.conf Regards, Chris 2008/9/18 Gergo Csibra [EMAIL PROTECTED]: Hello asterisk-users, My SIP phones are in pickupgroup, and if some of them ringing from other phone can pick up with *8 as usual. But I want to know if this happen. I've tried the a extension, but seems not working. Any other idea? -- Best regards, Gergo mailto:[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium training course
I agree with the training course, it takes extensive resources. But people that have been on in the ground floor should get a dCAP. I specifically said I was thread jacking, so possibly frowned upon, it is still on-topic. Finally, last I knew, you could go stand-by for the dCAP exam and not take the Asterisk Training classes. My experience with training classes (paper mills) gives a little scratch into the world of Asterisk. How many days is it? I would expect weeks if you are bright and months if you are't. I suggest that you self study, you will have a much better idea of why things work, don't work, and the things that should work but don't. Thanks, Steve Totaro 1.888.777.1888 On Wed, Sep 17, 2008 at 11:25 PM, Pascal Bruno [EMAIL PROTECTED] wrote: That is good you have all those years of experiences and you might know more than the instructor. But I dont see the connection, or the point you are trying to make. The question is that there is a space to apply a coupon code, and I was wondering how and where one could get one. I don't recall asking for free training, so I don't see why you are saying for that matter you think people with experience should get the dCAP. It doesn't make any sense to me. On Wed, Sep 17, 2008 at 10:42 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote: Anybody knows how to get a Coupon Code for the discount on the Asterisk training classes??? I am interested on taking that upcoming Asterisk Advance course, and 3K is kinda steep and considering I am still a college student paying this training out of my pocket, every bit helps. Sorry to thread jack. For that matter, I think old timers like myself should automatically get a dCAP. Six or seven years of Asterisk extensive experience should grandfather the dCAP and maybe even the training. I am sure I have a few tricks up my sleeve that the instructors don't know. If memory serves me correctly, there was talk about this very issue when the training and dCAP track came out. I will google it later. Thanks Steve Totaro 1.888.777.1888 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 482 Loop Detected
Hi, I am trying to establish a call between two users (A and B) but because I use Asterisk only to provide services, the request has to pass by the same Asterisk twice. Here what I am expecting to do : User A Equipment1 Asterisk1 Equipment1 Asterisk1 Equipment1 User B But when my request arrive in Asterisk the second time, Asterisk send me a 482 Loop Detected (because it was the sender of the last SIP session request). I would like to know if it is possible to disable loop detection in Asterisk or if there a way which could help me to solve this problem. Regards. -- Remi Druilhe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Info events
Hi Robb, Have a look in your features.conf file and see what keys you have enabled for transfers. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BRI or PRI callerid
Hi, I try to get anonymous calling working on ZAP. But I am unsuccessful on PRI as well as on BRI. I tried all parameters from the application SetCallerPres(). Nothing worked. I even traced with my ISP and they told me that I am not sending any parameter to hide the callerid. I found on the internet articles and mailinglist posts dating from 2003 that did not really help me. Im on a recent asterisk 1.4 from SVN and using euroisdn. Can anyone help? Is there a way to sniff/trace zap channels in an asterisk independent way? Best regards, Loic Didelot. -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Get rid of Really destroying SIP dialog
Hi, Whatever the verbosity level (even 0), my Asterisk console is full of Really destroying SIP dialog messages. Is there a way to get rid of those ? If not, do you think it deserves to marked as a bug ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Verbosity best practice
Hello, When managing a stable system, which verbosity level do you adopt ? Leaving a higher level helps to catch root cause, if for any reason, things go wrong. Leaving a lower level saves resources if you need (have) to backup logs. What are current best practices ? Do you change verbosity level during system lifecycle ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use
On Wednesday 17 September 2008 09:18:58 pm hugolivude wrote: I think it's better to find out what is listening on port 4520. CentOS 5 Asterisk 1.4.20 Presumably my other Asterisk server is listening on 4520. The problem here is that I can change the port, and it will work... until I reboot. When I reboot the problem reappears and I can fix it by changing the port again. Any other thoughts? what else on THIS machine is uusing port 4520? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI or PRI callerid
I managed to achieve that on a PRI line with the following: 1. On zapata.conf, for the PRI line channels, add facilityenable=yes usecallerid=yes usecallingpres=yes I do not known if these are all strictly required for anonymous calling, but it works for me. 2. On your extensions.conf, just prior do the Dial application invoke SetCallerPres(prohib_not_screened) 3. Your provider must also enable the apropriate functionalities on the PRI line. I believe they call IT CLIR (Calling Line Identification Restriction). Jorge Nunes Loic Didelot wrote: Hi, I try to get anonymous calling working on ZAP. But I am unsuccessful on PRI as well as on BRI. I tried all parameters from the application SetCallerPres(). Nothing worked. I even traced with my ISP and they told me that I am not sending any parameter to hide the callerid. I found on the internet articles and mailinglist posts dating from 2003 that did not really help me. Im on a recent asterisk 1.4 from SVN and using euroisdn. Can anyone help? Is there a way to sniff/trace zap channels in an asterisk independent way? Best regards, Loic Didelot. -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- PDMFC - Email: [EMAIL PROTECTED]; Web: http://www.pdmfc.com Phone: +351-213.572.029; Fax: +351-213.572.031 Address: Avenida Conde Valbom 30, 3 - 1050-068 Lisboa - Portugal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI or PRI callerid
And do You have usecallingpres=yes in your zapata.conf ? Hi,I try to get anonymous calling working on ZAP. But I am unsuccessful onPRI as well as on BRI. I tried all parameters from the application SetCallerPres(). Nothingworked. I even traced with my ISP and they told me that I am not sending anyparameter to hide the callerid. I found on the internet articles and mailinglist posts dating from 2003that did not really help me. Im on a recent asterisk 1.4 from SVN and using euroisdn. Can anyone help? Is there a way to sniff/trace zap channels in anasterisk independent way? Best regards,Loic Didelot. -- Loïc DIDELOTMIXvoip S.a.Tel: +352 20 20Fax: +352 20 [EMAIL PROTECTED]://www.mixvoip.com ___-- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, ArizonaRegister Now: http://www.astricon.net asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use
On Thu, Sep 18, 2008 at 05:31:08AM -0500, Anthony Messina wrote: On Wednesday 17 September 2008 09:18:58 pm hugolivude wrote: I think it's better to find out what is listening on port 4520. CentOS 5 Asterisk 1.4.20 Presumably my other Asterisk server is listening on 4520. The problem here is that I can change the port, and it will work... until I reboot. When I reboot the problem reappears and I can fix it by changing the port again. Any other thoughts? what else on THIS machine is uusing port 4520? Run (as root) netstat -lntup -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI or PRI callerid
Yes, thats why I do not get it. Also on BRI I know that it worked on my customers old PBX so I really exclude the carrier. Loic On Thu, 2008-09-18 at 12:38 +0200, Igor Zamocky wrote: And do You have usecallingpres=yes in your zapata.conf ? Hi,I try to get anonymous calling working on ZAP. But I am unsuccessful onPRI as well as on BRI. I tried all parameters from the application SetCallerPres(). Nothingworked. I even traced with my ISP and they told me that I am not sending anyparameter to hide the callerid. I found on the internet articles and mailinglist posts dating from 2003that did not really help me. Im on a recent asterisk 1.4 from SVN and using euroisdn. Can anyone help? Is there a way to sniff/trace zap channels in anasterisk independent way? Best regards,Loic Didelot. -- Loïc DIDELOTMIXvoip S.a.Tel: +352 20 20Fax: +352 20 [EMAIL PROTECTED]://www.mixvoip.com ___-- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, ArizonaRegister Now: http://www.astricon.net asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rxfax and txfax
Hi all, I want to configure my asterisk for sending and receiving faxes. I see in my sip.conf that i have to enable the t.38 capability. I have done that but the rxfax and txfax applications are not installed. They are not listed in applications when i do make menuselect. i have searched in voip-info wiki, found a pagehttp://www.voip-info.org/wiki/index.php?page_id=2583tk=99b8d086f0f28f4c1542comments_page=1but the links given on that page for downloading the applications are not working. I am using asterisk 1.4.2, i thaught the missing applications maybe included in latest version of asterisk but they are not, already downloaded and checked in asterisk 1.4.21. How can i install these applications. Are there anyother components required to make my asterisk a fax-passthru system. -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to make a Outgoing Call from Asterisk ?
Dear All, Pl. any one can give me help. B'coz I have to implicitly work for Outgoing call from PSTN Agent. I have also may to call out side the office from exten = s,n,Dial(Zap/4/111,60) on testing purpose. But, how to dial number via PSTN agent's phone like zero or nine dialing. I mean how to get dialtone before the dial and how to enter call number from PSTN agent's phone. -- With Regards, Hiren Mistry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verbosity best practice
Its a good question I have lots of disk space so leave it high, I would rather have the detail if I need it It probably would seem sensible to revisit stable systems after a year and lower the verbosity, but then since I can afford the space I am not too fussed. Cheers Duncan Olivier wrote: Hello, When managing a stable system, which verbosity level do you adopt ? Leaving a higher level helps to catch root cause, if for any reason, things go wrong. Leaving a lower level saves resources if you need (have) to backup logs. What are current best practices ? Do you change verbosity level during system lifecycle ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pre-paid Billing
Hi Guys, we need an urgent help with Pre-paid Billing. We are using Asterisk at work with our own prepaid billing system. We calculate max number of minutes user is allowed to talk based on his balance and destination. We then used Dial command with S(x) parameter to create a call. However, this is a problem when user makes multiple calls simulatenously. What is the best way to handle it. Any suggestions. Please do not answer suggestion some billing packages as we have our own billing system which we need to enhance. Technical answer on howto do with bare asterisk or algorithm would help. Jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rxfax and txfax
On Thursday 18 September 2008 13:34:06 Rizwan Hisham wrote: How can i install these applications. Are there anyother components required to make my asterisk a fax-passthru system. http://sourceforge.net/projects/agx-ast-addons t. -- knowledgeTools® ... managing complexity. -- knowledgeTools International GmbH Wallstraße 15 / 15 a 10179 Berlin Fon: +49 30 726 169 20 Fax: +49 30 726 169 249 [EMAIL PROTECTED] www.knowledgetools.de Sitz Berlin, AG Berlin-Charlottenburg, HRB 86378 Geschäftsführer: Oliver Seyboldt, Reinhard Kunz -- This eMail communication (and any attachment/s) may contain confidential or privileged information and is intended only for the individual(s) or entity named above and to others who have been specifically authorized to receive it. If you are not the intended recipient, please do not read, copy, use or disclose the contents of this communication to others. Please notify the sender that you have received this e-mail in error by reply e-mail, and delete the e-mail subsequently. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Info events
On Sat, 2008-09-13 at 01:13 +0100, robb wrote: I'm trying to get a simens IP pjone working so I can transfer calls using the recall key when I run sip debug I get the below text on screen, but I don't get dialtone returned, any advice would be greatly appriciated I don't claim to know much about the SIP protocol, but it's my (possibly flawed) understanding that Asterisk doesn't interpret an incoming FLASH message as part of a SIP INFO packet as a request to put the call on hold or to transfer the call. And just to be perfectly clear, I personally don't think that FLASH is the proper way to initiate these sorts of events in the SIP protocol. To do a call transfer in SIP, for example, you should be using the REFER primative. (And just for the sake of general trivia, I haven't yet been able to figure out how to get Asterisk to generate an outbound FLASH message over SIP INFO either.) -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Restrict SIP registration to one ip address only?
On Wed, 17 Sep 2008, Jared Smith wrote: On Wed, 2008-09-17 at 19:58 +0200, Remco Barendse wrote: Why doesn't Asterisk allow both usernamepass as well as setting an ip adress on a sip.extension? It does. To enforce ACLs on a SIP user or peer or friend, simply use permit and deny statements to allow and disallow various IP addresses or subnets. Standard practice seems to be to deny everything first, then specifically allow other IP addresses. [user] type=friend secret=mypassword host=dynamic deny=0.0.0.0/0 permit=10.1.2.3 permit=192.168.123.0/24 permit=192.168.222.0/255.255.255.0 Cool, this is exactly what i was looking for, i couldn't find a reference to it anywhere else. Suprising that this feature isn't used much, i would suspect that many asterisk installations (including mine) have very simple (short) extension numbers which makes brute forcing them rather easy. I was never concerned about short extension numbers and easy passwords until the need came up to connect to my * box from outside. Thanks again! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verbosity best practice
2008/9/18 Duncan Turnbull [EMAIL PROTECTED] Its a good question I have lots of disk space so leave it high, I would rather have the detail if I need it It probably would seem sensible to revisit stable systems after a year and lower the verbosity, but then since I can afford the space I am not too fussed. Cheers Duncan Once, a customer of mine asked one employee's calls listing. When I read CDR, I discovered most of it was unusable, due to a mistake in dialplan. Then, I was very happy to use logs as a second source of CDR : I could parse logs to complement CDR and provide the listing I was asked. At that time, I told myself I should think it over and elaborate some rule about logging verbosity. So at the moment, I told myself I would never set verbosity any lower to the point you wouldn't be able to rebuild CDR from it. Another thought is the other day, when I tried to shrink customer's logs, after 30mn of processing, I got temperature is becoming too hot warning in syslog. So I didn't take any chance and stopped the ongoing job. So having plenty (too much) of logs has a price as I couldn't save them as conveniently as I would have thought. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP-XR
Another question :
exten = 999,n,Log(DEBUG,local_ssrc:
${CHANNEL(rtpqos,audio,local_ssrc)})
Are those ${CHANNEL( (rtpqos,audio,local_ssrc)} values available today in an
Asterisk version or is it something describing what should be coded ?
Regards
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[asterisk-users] CHANNEL((rtpqos|audio|local_ssrc)) (was: Re: RTCP-XR)
Olivier schrieb:
Another question :
exten = 999,n,Log(DEBUG,local_ssrc:
${CHANNEL(rtpqos,audio,local_ssrc)})
Are those ${CHANNEL( (rtpqos,audio,local_ssrc)} values available today in an
Asterisk version
Yes. 1.4 and 1.6. But only for SIP channels obviously.
chan_sip.c: acf_channel_read()
Philipp Kempgen
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Re: [asterisk-users] Help with MFC/R2
Dae, Activate debug full: asterisk -vr in other console do: tail -vf /var/log/asterisk/full Try to put call and send us more details about your logs Regards, Luis Morales On Thu, Sep 18, 2008 at 11:49 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: In fact I see 1101 in the rx bits on all channels... But I have in parallel one old Panasonic Key Phone system (Actually in production, to be replaced by asterisk), and it's works perfectly and immediately once I pass the E1 cables to there... So, the problem is not from Telco... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Wednesday, September 17, 2008 10:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 It seems to me your lines are blocked. Execute zttool and if you see 1101 in the rx bits, it means the telco (or whatever you have in the other end) has blocked their side. If this is a telco line you need to call them and tell them to unblock your lines. On Wed, Sep 17, 2008 at 10:33 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Thank you for the reply I shutdown asterisk and tried again and I have to following logs... OUTGOING TEST : Testcall.conf caller yes destination-no 6055151 originating-no 7309130 protocol-class mfcr2 protocol-variant ar,20,4 circuits 1-2 Log: ./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309130' to '6055151' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309131' to '6055152' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) MFC/R2 Chan 2: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Incoming test : Testcall.conf caller no protocol-class mfcr2 protocol-variant ar,20,4 on-offered answer circuits 1-2 Log: Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: local_unblocking_expired MFC/R2 Chan 1: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Main thread Main thread Seems no any response from far side... Do you have any ideas?? Only one time, I got the following log: #./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] Chan 2: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 1: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 2: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 1: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern MFC/R2 Chan 2: local_unblocking_expired MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread MFC/R2 Chan 2: - 1101 [1/BLOCKED /Idle /Idle ] Chan 2: -- Far end blocked! :-( Chan 2: -- Far end blocked! :-( MFC/R2 Chan 1: - 1101 [1/BLOCKED /Idle /Idle ] Chan 1: -- Far end blocked! :-( Chan 1: -- Far end blocked! :-( Main thread Main thread Main thread But after rerunning the test, I only get the first log (w/o Far end replies.) Any help will be really appreciated! Thank you! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Wednesday,
Re: [asterisk-users] Help with MFC/R2
Do as Luis says, however, I feel that as long you keep getting 1101 Unicall won't work. AFAIK The only IDLE bit pattern recognized by libmfcr2 as IDLE is 10XX, as long you have 11 in the first 2 bits (AB), libmfcr2 will report the lines as blocked. On Thu, Sep 18, 2008 at 8:16 AM, Luis Morales [EMAIL PROTECTED] wrote: Dae, Activate debug full: asterisk -vr in other console do: tail -vf /var/log/asterisk/full Try to put call and send us more details about your logs Regards, Luis Morales On Thu, Sep 18, 2008 at 11:49 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: In fact I see 1101 in the rx bits on all channels... But I have in parallel one old Panasonic Key Phone system (Actually in production, to be replaced by asterisk), and it's works perfectly and immediately once I pass the E1 cables to there... So, the problem is not from Telco... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Wednesday, September 17, 2008 10:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 It seems to me your lines are blocked. Execute zttool and if you see 1101 in the rx bits, it means the telco (or whatever you have in the other end) has blocked their side. If this is a telco line you need to call them and tell them to unblock your lines. On Wed, Sep 17, 2008 at 10:33 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Thank you for the reply I shutdown asterisk and tried again and I have to following logs... OUTGOING TEST : Testcall.conf caller yes destination-no 6055151 originating-no 7309130 protocol-class mfcr2 protocol-variant ar,20,4 circuits 1-2 Log: ./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309130' to '6055151' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309131' to '6055152' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) MFC/R2 Chan 2: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Incoming test : Testcall.conf caller no protocol-class mfcr2 protocol-variant ar,20,4 on-offered answer circuits 1-2 Log: Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: local_unblocking_expired MFC/R2 Chan 1: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Main thread Main thread Seems no any response from far side... Do you have any ideas?? Only one time, I got the following log: #./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] Chan 2: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 1: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 2: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 1: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern MFC/R2 Chan 2: local_unblocking_expired MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread MFC/R2 Chan 2: - 1101 [1/BLOCKED /Idle /Idle ] Chan 2: -- Far end blocked! :-( Chan 2: -- Far end blocked! :-( MFC/R2 Chan 1: - 1101 [1/BLOCKED /Idle /Idle ] Chan 1: -- Far end blocked! :-( Chan 1: -- Far
Re: [asterisk-users] Pre-paid Billing
You need some outside process to keep call state, probably using the Manager API and/or AGI. The outside process can listen to periodic call setup events at a relatively low polling interval and make appropriate adjustments to the user's credit in the database, which will then allow you to determine whether to allow another call to be set up given the available credit, etc. to a relatively high degree of accuracy. Jim Boykin wrote: Hi Guys, we need an urgent help with Pre-paid Billing. We are using Asterisk at work with our own prepaid billing system. We calculate max number of minutes user is allowed to talk based on his balance and destination. We then used Dial command with S(x) parameter to create a call. However, this is a problem when user makes multiple calls simulatenously. What is the best way to handle it. Any suggestions. Please do not answer suggestion some billing packages as we have our own billing system which we need to enhance. Technical answer on howto do with bare asterisk or algorithm would help. Jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pre-paid Billing
Isn't 'don't allow multiple calls' acceptable solution? At least, it's the simplest one :) I can imagine solution with multiple calls allowed, but it needs some external synchronous processing. With every call you should start process, that will decrement user's balance based on dialled destination, you have to update balance every second. After balance=0 you just kill active call(s). The fact, that there are multiple calls means nothing, just more processes will decrement balance for the same user. Btw, this will give You oportunity upgrade balance during call, so active call can be longer than we originally thought - of course, you should not use S(x). There will be probably a lot of other / more effective, easier, ... / ideas :) Igor Hi Guys, we need an urgent help with Pre-paid Billing. We are using Asterisk at work with our own prepaid billing system. We calculate max number of minutes user is allowed to talk based on his balance and destination. We then used Dial command with S(x) parameter to create a call. However, this is a problem when user makes multiple calls simulatenously. What is the best way to handle it. Any suggestions. Please do not answer suggestion some billing packages as we have our own billing system which we need to enhance. Technical answer on howto do with bare asterisk or algorithm would help. Jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with MFC/R2
Hello I got: [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call called - 'g1/6055151' [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call caller id - '1102' [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: no UC_CATEGORY specified for chan UniCall/1-1, using default NATIONAL_SUBSCRIBER [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Call control(1) [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Make call [Sep 17 19:24:50] WARNING[4934] chan_unicall.c: Make call failed - Blocked [Sep 17 19:24:50] VERBOSE[4934] logger.c: -- Couldn't call g1/6055151 [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Channel gains [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Channel switching [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: Hangup: channel: 1 index = 0, normal = 11, callwait = -1, thirdcall = -1 [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: Updated conferencing on 1, with 0 conference users [Sep 17 19:24:50] VERBOSE[4934] logger.c: -- Hungup 'UniCall/1-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Thursday, September 18, 2008 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Dae, Activate debug full: asterisk -vr in other console do: tail -vf /var/log/asterisk/full Try to put call and send us more details about your logs Regards, Luis Morales On Thu, Sep 18, 2008 at 11:49 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: In fact I see 1101 in the rx bits on all channels... But I have in parallel one old Panasonic Key Phone system (Actually in production, to be replaced by asterisk), and it's works perfectly and immediately once I pass the E1 cables to there... So, the problem is not from Telco... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Wednesday, September 17, 2008 10:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 It seems to me your lines are blocked. Execute zttool and if you see 1101 in the rx bits, it means the telco (or whatever you have in the other end) has blocked their side. If this is a telco line you need to call them and tell them to unblock your lines. On Wed, Sep 17, 2008 at 10:33 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Thank you for the reply I shutdown asterisk and tried again and I have to following logs... OUTGOING TEST : Testcall.conf caller yes destination-no 6055151 originating-no 7309130 protocol-class mfcr2 protocol-variant ar,20,4 circuits 1-2 Log: ./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309130' to '6055151' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309131' to '6055152' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) MFC/R2 Chan 2: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Incoming test : Testcall.conf caller no protocol-class mfcr2 protocol-variant ar,20,4 on-offered answer circuits 1-2 Log: Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: local_unblocking_expired MFC/R2 Chan 1: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Main thread Main thread Seems no any response from far side... Do you have any ideas?? Only one time, I got the following log: #./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan
Re: [asterisk-users] CHANNEL((rtpqos|audio|local_ssrc)) (was: Re: RTCP-XR)
2008/9/18 Philipp Kempgen [EMAIL PROTECTED]
Olivier schrieb:
Another question :
exten = 999,n,Log(DEBUG,local_ssrc:
${CHANNEL(rtpqos,audio,local_ssrc)})
Are those ${CHANNEL( (rtpqos,audio,local_ssrc)} values available today in
an
Asterisk version
Yes. 1.4 and 1.6. But only for SIP channels obviously.
chan_sip.c: acf_channel_read()
Thanks !
Philipp Kempgen
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Re: [asterisk-users] Pre-paid Billing
Igor Zamocky wrote: Isn't 'don't allow multiple calls' acceptable solution? At least, it's the simplest one :) I can imagine solution with multiple calls allowed, but it needs some external synchronous processing. With every call you should start process, that will decrement user's balance based on dialled destination, you have to update balance every second. After balance=0 you just kill active call(s). The fact, that there are multiple calls means nothing, just more processes will decrement balance for the same user. Btw, this will give You oportunity upgrade balance during call, so active call can be longer than we originally thought - of course, you should not use S(x). There will be probably a lot of other / more effective, easier, ... / ideas :) You don't have to update the balance every second - increments of something like 10 seconds will do. And you can have one synchronous process - not many - that listens to Manager events and updates the call times and balances accordingly. An outside process can also trigger a Hangup event causing the call to be hung up if credit is exhausted, or too low. Then, you can define a minimum formula for the balance required to admit a new call. Something like a minute of credit being required after subtracting the usage of all existing simultaneous calls in progress at the next projected utilisation polling interval. But you are essentially correct. Things are, of course, far easier if you just don't allow multiple simultaneous calls. :-) -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Custom Voicemail emails
So here is the deal. I have an Asterisk server here at work that I have recently taken over and the boss is wanting the server to do a lot of things that it didn't do before. I have already configured much of what he wanted including a voice messaging line where anyone can call in and leave a message and then he would get that message in his email. However, the boss wants his email subject to read something like This is an urgent message through the HISG voice messaging system so he knows that that message came through that number as opposed to his voicemail box that already gets forwarded there. The default is the [PBX]: New Message 10 in mailbox 0307. At second glance he would know which voicemail box is his line but he wants things to be different and so I am trying to make that happen. I know there is the 'emailsubject' option. I haven't tried this yet but my concern is that it will set the subject the same on every single box (obviously what the command is designed for). I can I customize a voicemail message so that if something comes in on our 0307 line it has a certain message and then we might get a message on 1942 line that we want a different subject. I am new to Asterisk, only been messing with it for a couple of weeks. Any thoughts? __ Steve Anness ICT Support Specialist Humanitarian International Services Group ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Restrict SIP registration to one ip address only?
Remco Barendse schrieb: Suprising that this feature isn't used much, i would suspect that many asterisk installations (including mine) have very simple (short) extension numbers which makes brute forcing them rather easy. Extension numbers and SIP account basically have nothing to do with each other. If you name your SIP accounts after the respective extension number, you have a security issue in your design which you should solve first! A SIP peer definition can be like [Remcossoftclientathislaptop] type=friend secret=verysecretpassword ... And then in the diaplan you just do something like [internalcontext] exten = 10,1,Dial(SIP/Remcossoftclientathislaptop,30) exten = 10,2,Hangup() ... So, the username for you SIP client would be Remcossoftclientathislaptop while the dialled extension would be 10. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - How to stream a A-Law/wav file to a browser ?
Hi, How can I create a web page allowing people to listen (with their own PC) a couple of .wav/a-law files stored on a Linux server ? Chances are users would access this web page from Internet Explorer but if I could make it available to other browsers, that would be better. I googled a bit and couldn't find a tag such as media://myaudiofile.wav that would fulfill this spec. As much as possible, I would be happy to avoid configuring browser plugins and so on. So if this media:// could be already installed and running in users PCs, that would be fine. I've read Red5 servers/Flash players combination could respond but I'm not too confident about a-law support and Red5 installation complexity for a 100% pure beginner. What do you think ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with MFC/R2
Ok, in your E1 setup: 1-15: to outgoing calls 16-30: for incomming calls ? Now for make calls your telephone company must be provide MFC-R2 signaling. In your case the logs files show an invalid signal on make call. Regards, Luis Morales On Fri, Sep 19, 2008 at 10:10 AM, Dae Yeung Um [EMAIL PROTECTED] wrote: Hello I got: [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call called - 'g1/6055151' [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call caller id - '1102' [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: no UC_CATEGORY specified for chan UniCall/1-1, using default NATIONAL_SUBSCRIBER [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Call control(1) [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Make call [Sep 17 19:24:50] WARNING[4934] chan_unicall.c: Make call failed - Blocked [Sep 17 19:24:50] VERBOSE[4934] logger.c: -- Couldn't call g1/6055151 [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Channel gains [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Channel switching [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: Hangup: channel: 1 index = 0, normal = 11, callwait = -1, thirdcall = -1 [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: Updated conferencing on 1, with 0 conference users [Sep 17 19:24:50] VERBOSE[4934] logger.c: -- Hungup 'UniCall/1-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Thursday, September 18, 2008 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Dae, Activate debug full: asterisk -vr in other console do: tail -vf /var/log/asterisk/full Try to put call and send us more details about your logs Regards, Luis Morales On Thu, Sep 18, 2008 at 11:49 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: In fact I see 1101 in the rx bits on all channels... But I have in parallel one old Panasonic Key Phone system (Actually in production, to be replaced by asterisk), and it's works perfectly and immediately once I pass the E1 cables to there... So, the problem is not from Telco... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Wednesday, September 17, 2008 10:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 It seems to me your lines are blocked. Execute zttool and if you see 1101 in the rx bits, it means the telco (or whatever you have in the other end) has blocked their side. If this is a telco line you need to call them and tell them to unblock your lines. On Wed, Sep 17, 2008 at 10:33 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Thank you for the reply I shutdown asterisk and tried again and I have to following logs... OUTGOING TEST : Testcall.conf caller yes destination-no 6055151 originating-no 7309130 protocol-class mfcr2 protocol-variant ar,20,4 circuits 1-2 Log: ./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309130' to '6055151' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309131' to '6055152' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) MFC/R2 Chan 2: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Incoming test : Testcall.conf caller no protocol-class mfcr2 protocol-variant ar,20,4 on-offered answer circuits 1-2 Log: Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: local_unblocking_expired MFC/R2 Chan 1: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Main thread Main thread Seems no any response from far side... Do you have any ideas?? Only one time, I got the following log: #./testcall Chan 1, class 'mfcr2', variant 'ar,20,4',
Re: [asterisk-users] Pre-paid Billing
Thanks guys for inputs...not allowing multiple call is not an option - essentional thats the problem we try to solve :) Since we have our own CDR module, we can avoid external process. What are the evens to listen for? Other ideas will also be appreciated. On Thu, Sep 18, 2008 at 8:23 PM, Alex Balashov [EMAIL PROTECTED] wrote: Igor Zamocky wrote: Isn't 'don't allow multiple calls' acceptable solution? At least, it's the simplest one :) I can imagine solution with multiple calls allowed, but it needs some external synchronous processing. With every call you should start process, that will decrement user's balance based on dialled destination, you have to update balance every second. After balance=0 you just kill active call(s). The fact, that there are multiple calls means nothing, just more processes will decrement balance for the same user. Btw, this will give You oportunity upgrade balance during call, so active call can be longer than we originally thought - of course, you should not use S(x). There will be probably a lot of other / more effective, easier, ... / ideas :) You don't have to update the balance every second - increments of something like 10 seconds will do. And you can have one synchronous process - not many - that listens to Manager events and updates the call times and balances accordingly. An outside process can also trigger a Hangup event causing the call to be hung up if credit is exhausted, or too low. Then, you can define a minimum formula for the balance required to admit a new call. Something like a minute of credit being required after subtracting the usage of all existing simultaneous calls in progress at the next projected utilisation polling interval. But you are essentially correct. Things are, of course, far easier if you just don't allow multiple simultaneous calls. :-) -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pre-paid Billing
Take a look at the Asterisk Manager API documentation on voip-info.org and experiment empirically by connecting and watching what transpires. On Thu, September 18, 2008 11:14 am, Jim Boykin wrote: Thanks guys for inputs...not allowing multiple call is not an option - essentional thats the problem we try to solve :) Since we have our own CDR module, we can avoid external process. What are the evens to listen for? Other ideas will also be appreciated. On Thu, Sep 18, 2008 at 8:23 PM, Alex Balashov [EMAIL PROTECTED] wrote: Igor Zamocky wrote: Isn't 'don't allow multiple calls' acceptable solution? At least, it's the simplest one :) I can imagine solution with multiple calls allowed, but it needs some external synchronous processing. With every call you should start process, that will decrement user's balance based on dialled destination, you have to update balance every second. After balance=0 you just kill active call(s). The fact, that there are multiple calls means nothing, just more processes will decrement balance for the same user. Btw, this will give You oportunity upgrade balance during call, so active call can be longer than we originally thought - of course, you should not use S(x). There will be probably a lot of other / more effective, easier, ... / ideas :) You don't have to update the balance every second - increments of something like 10 seconds will do. And you can have one synchronous process - not many - that listens to Manager events and updates the call times and balances accordingly. An outside process can also trigger a Hangup event causing the call to be hung up if credit is exhausted, or too low. Then, you can define a minimum formula for the balance required to admit a new call. Something like a minute of credit being required after subtracting the usage of all existing simultaneous calls in progress at the next projected utilisation polling interval. But you are essentially correct. Things are, of course, far easier if you just don't allow multiple simultaneous calls. :-) -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?
How do you feel about converting them to RIFF/MSPCM WAV format and encoding them into MP3? On Thu, September 18, 2008 11:01 am, Olivier wrote: Hi, How can I create a web page allowing people to listen (with their own PC) a couple of .wav/a-law files stored on a Linux server ? Chances are users would access this web page from Internet Explorer but if I could make it available to other browsers, that would be better. I googled a bit and couldn't find a tag such as media://myaudiofile.wav that would fulfill this spec. As much as possible, I would be happy to avoid configuring browser plugins and so on. So if this media:// could be already installed and running in users PCs, that would be fine. I've read Red5 servers/Flash players combination could respond but I'm not too confident about a-law support and Red5 installation complexity for a 100% pure beginner. What do you think ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium training course
On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote: On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote: Anybody knows how to get a Coupon Code for the discount on the Asterisk training classes??? I am interested on taking that upcoming Asterisk Advance course, and 3K is kinda steep and considering I am still a college student paying this training out of my pocket, every bit helps. Sorry to thread jack. For that matter, I think old timers like myself should automatically get a dCAP. Six or seven years of Asterisk extensive experience should grandfather the dCAP and maybe even the training. I am sure I have a few tricks up my sleeve that the instructors don't know. If memory serves me correctly, there was talk about this very issue when the training and dCAP track came out. I will google it later. Nobody, including Mark Spencer and myself, have gotten a free pass on the dCAP. That said, I think you may be able to take the test for cheap. I don't know the exact price, but in any bootcamp, they allow people to come down on the last day and take only the test, as I did, if they have the space and resources (specifically, for the practical portion). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get rid of Really destroying SIP dialog
On Thursday 18 September 2008 05:16:21 Olivier wrote: Whatever the verbosity level (even 0), my Asterisk console is full of Really destroying SIP dialog messages. Is there a way to get rid of those ? Turn off debugging: core set debug 0 (and don't specify -d on your command line). If not, do you think it deserves to marked as a bug ? It is a debugging message, not a bug. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Voicemail emails
On Thursday 18 September 2008 09:54:39 Steve Anness wrote: So here is the deal. I have an Asterisk server here at work that I have recently taken over and the boss is wanting the server to do a lot of things that it didn't do before. I have already configured much of what he wanted including a voice messaging line where anyone can call in and leave a message and then he would get that message in his email. However, the boss wants his email subject to read something like This is an urgent message through the HISG voice messaging system so he knows that that message came through that number as opposed to his voicemail box that already gets forwarded there. The default is the [PBX]: New Message 10 in mailbox 0307. At second glance he would know which voicemail box is his line but he wants things to be different and so I am trying to make that happen. I know there is the 'emailsubject' option. I haven't tried this yet but my concern is that it will set the subject the same on every single box (obviously what the command is designed for). I can I customize a voicemail message so that if something comes in on our 0307 line it has a certain message and then we might get a message on 1942 line that we want a different subject. Currently, there is no such capability. Coding it would not be very difficult, however. I would suggest using the per-mailbox settings and simply adding the option to code an email subject per mailbox, and then default to the generic subject, if one is not otherwise specified in the mailbox. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?
On Thu, 18 Sep 2008, Olivier wrote:
Hi,
How can I create a web page allowing people to listen (with their own PC) a
couple of .wav/a-law files stored on a Linux server ?
Chances are users would access this web page from Internet Explorer but if I
could make it available to other browsers, that would be better.
I googled a bit and couldn't find a tag such as media://myaudiofile.wav that
would fulfill this spec.
If the web server is running php, then this will work:
?
$action = $HTTP_GET_VARS[action] ;
$file = $HTTP_GET_VARS[file] ;
$caller = $HTTP_GET_VARS[caller] ;
if (empty ($action) || empty ($file))
die (Something went wrong) ;
// Open the file
$fileName = /prefix/ . $file ;
$fd = @fopen ($fileName, rb) ;
if ($fd === FALSE)
{ Header (Location: . $caller . .php?error=1) ; die () ; }
// Send the headers to the browser
$len = filesize ($fileName) ;
Header (Accept-Ranges: bytes);
Header (Content-Length: $len) ;
Header (Keep-Alive: timeout=2, max=100) ;
Header (Connection: Keep-Alive) ;
Header (Content-Type: audio/x-wav) ;
if ($action == download)
{
Header (Content-Disposition: attachment; filename=\$fileName\);
Header (Content-Description: File Transfer);
}
// Transmit the file in 8K blocks
while (!feof ($fd) (connection_status () == 0))
{
set_time_limit (0) ;
print (fread ($fd, 1024*8)) ;
flush () ;
}
fclose ($fd) ;
?
If this was called playback.php, then you'd reference it in other HTML
code with:
a href=payback.php?action=playfile=music.wavcaller=thisFileClick
here to play/a
I've found that seems to let most browsers play (or download) most audio
files, (most of the time ;-)
The browser will (should) do whatever it's configured to do with audio
files. I use this to let people playback voicemail and call recordings.
Gordon
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[asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?
Hi, It has IOS is 12.3(14)T7 - I'm wondering if this router can be made SIP aware. Apparently, this current firmware/programming is not, one way audio problems. Is there a version that support VoIP directly for this router? Thanks, Bart___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium training course
Tilghman Lesher wrote: On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote: On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote: Anybody knows how to get a Coupon Code for the discount on the Asterisk training classes??? I am interested on taking that upcoming Asterisk Advance course, and 3K is kinda steep and considering I am still a college student paying this training out of my pocket, every bit helps. Sorry to thread jack. For that matter, I think old timers like myself should automatically get a dCAP. Six or seven years of Asterisk extensive experience should grandfather the dCAP and maybe even the training. I am sure I have a few tricks up my sleeve that the instructors don't know. If memory serves me correctly, there was talk about this very issue when the training and dCAP track came out. I will google it later. Nobody, including Mark Spencer and myself, have gotten a free pass on the dCAP. That said, I think you may be able to take the test for cheap. I don't know the exact price, but in any bootcamp, they allow people to come down on the last day and take only the test, as I did, if they have the space and resources (specifically, for the practical portion). I also used to think the old timers should get grandfathered dCAP cert. Then I looked at some of the stuff the dCAP certification tests for and realized that almost nobody out there that learned Asterisk from the docs and use it in a real world install would be able to pass the dCAP. As Tilghman said you can take JUST the dCAP test for a reasonable fee without having to take the classes. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?
Barton I think this will help you out http://articles.techrepublic.com.com...1-6136216.html http://articles.techrepublic.com.com/5100-1035_11-6136216.html Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers mailto:[EMAIL PROTECTED] , CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barton Fisher Sent: Thursday, September 18, 2008 12:21 PM To: asterisk-user Discussion Subject: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware? Hi, It has IOS is 12.3(14)T7 - I'm wondering if this router can be made SIP aware. Apparently, this current firmware/programming is not, one way audio problems. Is there a version that support VoIP directly for this router? Thanks, Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?
Gordon Henderson schrieb: If the web server is running php, then this will work: ? $action = $HTTP_GET_VARS[action] ; $file = $HTTP_GET_VARS[file] ; $caller = $HTTP_GET_VARS[caller] ; if (empty ($action) || empty ($file)) die (Something went wrong) ; // Open the file $fileName = /prefix/ . $file ; $fd = @fopen ($fileName, rb) ; Without any validation of the filename? It could be ../../secret/file. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?
2008/9/18 Alex Balashov [EMAIL PROTECTED] How do you feel about converting them to RIFF/MSPCM WAV format and encoding them into MP3? Why not ? I don't know why I came to stick with A-law (as this is the codec used elsewhere and audio prompts will be recorded using hardphone) but thinking over it now, you're right that this shouldn't be a requirement. If this simplifies streaming, it won't complexify recording and management. Thanks for pointing this. On Thu, September 18, 2008 11:01 am, Olivier wrote: Hi, How can I create a web page allowing people to listen (with their own PC) a couple of .wav/a-law files stored on a Linux server ? Chances are users would access this web page from Internet Explorer but if I could make it available to other browsers, that would be better. I googled a bit and couldn't find a tag such as media://myaudiofile.wav that would fulfill this spec. As much as possible, I would be happy to avoid configuring browser plugins and so on. So if this media:// could be already installed and running in users PCs, that would be fine. I've read Red5 servers/Flash players combination could respond but I'm not too confident about a-law support and Red5 installation complexity for a 100% pure beginner. What do you think ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pre-paid Billing
Another idea can be have the customers to opt-in for auto-refill if they want to use multiple call feature. Usually this does not have be a high number, just autorefill the account if the balance goes down $1. Jai www.didforsale.com *Buy DID at low cost http://www.didforsale.com; On Thu, Sep 18, 2008 at 8:14 AM, Jim Boykin [EMAIL PROTECTED] wrote: Thanks guys for inputs...not allowing multiple call is not an option - essentional thats the problem we try to solve :) Since we have our own CDR module, we can avoid external process. What are the evens to listen for? Other ideas will also be appreciated. On Thu, Sep 18, 2008 at 8:23 PM, Alex Balashov [EMAIL PROTECTED] wrote: Igor Zamocky wrote: Isn't 'don't allow multiple calls' acceptable solution? At least, it's the simplest one :) I can imagine solution with multiple calls allowed, but it needs some external synchronous processing. With every call you should start process, that will decrement user's balance based on dialled destination, you have to update balance every second. After balance=0 you just kill active call(s). The fact, that there are multiple calls means nothing, just more processes will decrement balance for the same user. Btw, this will give You oportunity upgrade balance during call, so active call can be longer than we originally thought - of course, you should not use S(x). There will be probably a lot of other / more effective, easier, ... / ideas :) You don't have to update the balance every second - increments of something like 10 seconds will do. And you can have one synchronous process - not many - that listens to Manager events and updates the call times and balances accordingly. An outside process can also trigger a Hangup event causing the call to be hung up if credit is exhausted, or too low. Then, you can define a minimum formula for the balance required to admit a new call. Something like a minute of credit being required after subtracting the usage of all existing simultaneous calls in progress at the next projected utilisation polling interval. But you are essentially correct. Things are, of course, far easier if you just don't allow multiple simultaneous calls. :-) -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] server farm workarounds? (was Re: Restrict SIP registration to one ip address only?)
Apparently I mis-interpreted was the original poster was wanting. Good thing. I'm glad he has a solid answer. But, this does bring up the my issue of yore, and I'd be curious how people have handled this. Key items: * It's a distributed server farm. There are N asterisk servers serving hundreds of customers, each with dozens of extensions. For the sake of this example, I'll just pretend there are 3 asterisk boxes. * It's distributed. It's not a primary/failover. It should not matter which asterisk box takes an incoming call from the PSTN. Each server is independent and does not require that the other servers be operating. * Each asterisk box should be able to send a call to each SIP CPE (Polycom 550, etc.) directly. * Nearly every SIP CPE I have encountered requires that it register before making calls. Most support multiple registration profiles for redundancy, however, none of them will actually register with more than one server at the same time. * Nearly every SIP CPE I have encountered has a permissive mode of some kind that allows it to take calls from multiple IP addresses. * Network problems and CPE problems (such as people unplugging their phone) do happen, so it's important that the asterisk box know if a phone is up or not. (Timing out is problematic and takes a while.) And, the problem: * Asterisk does not seem to have a way to monitor an extension (peer) that might also be used to register. In other words, a registrable peer ignores the 'qualify=yes' setting. * Using two peer entries in sip.conf for the same CPE (one registrable and one static/qualified) creates some bad scenarios on transfers, conference calls, and other applications. Ideas? How have others gotten around this restriction? Ironically, all the problems would go away if a registrable peers could use qualify=yes. It's almost a bug. Thanks, John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?
2008/9/18 Gordon Henderson
[EMAIL PROTECTED][EMAIL PROTECTED]
On Thu, 18 Sep 2008, Olivier wrote:
Hi,
How can I create a web page allowing people to listen (with their own PC)
a
couple of .wav/a-law files stored on a Linux server ?
Chances are users would access this web page from Internet Explorer but
if I
could make it available to other browsers, that would be better.
I googled a bit and couldn't find a tag such as media://myaudiofile.wav
that
would fulfill this spec.
If the web server is running php,
You read in my mind : it will certainly run php !
then this will work:
?
$action = $HTTP_GET_VARS[action] ;
$file = $HTTP_GET_VARS[file] ;
$caller = $HTTP_GET_VARS[caller] ;
if (empty ($action) || empty ($file))
die (Something went wrong) ;
// Open the file
$fileName = /prefix/ . $file ;
$fd = @fopen ($fileName, rb) ;
if ($fd === FALSE)
{ Header (Location: . $caller . .php?error=1) ; die () ; }
// Send the headers to the browser
$len = filesize ($fileName) ;
Header (Accept-Ranges: bytes);
Header (Content-Length: $len) ;
Header (Keep-Alive: timeout=2, max=100) ;
Header (Connection: Keep-Alive) ;
Header (Content-Type: audio/x-wav) ;
if ($action == download)
{
Header (Content-Disposition: attachment; filename=\$fileName\);
Header (Content-Description: File Transfer);
}
// Transmit the file in 8K blocks
while (!feof ($fd) (connection_status () == 0))
{
set_time_limit (0) ;
print (fread ($fd, 1024*8)) ;
flush () ;
}
fclose ($fd) ;
?
If this was called playback.php, then you'd reference it in other HTML
code with:
a href=payback.php?action=playfile=music.wavcaller=thisFileClick
here to play/a
I've found that seems to let most browsers play (or download) most audio
files, (most of the time ;-)
The browser will (should) do whatever it's configured to do with audio
files. I use this to let people playback voicemail and call recordings.
Is that all ?
You just have to specify Content-Type: audio/x-wav and it will work with
an Internet Explorer browser ?
That's good news for me.
Thanks for sharing this.
Cheers
Gordon
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[asterisk-users] device probe order question
I have an Ubuntu 8 server running Asterisk 1.4 with three interface cards in it: * Wildcard TDM400P * Wildcard TDM410P * Wildcard TE122 I'm using zaptel 1.4.11, and the difficulty I'm running into is that with EVERY reboot, the order in which the hardware appears changes. This makes ztscan cough up a different order for the spans, which makes it nearly impossible to use the Digium GUI as after reboots, it complains that the hardware has changed. Is there a way to lock down probe order on boot, or somehow write zaptel.conf in such a way that port and span assignments never change between reboots? -- Jason T. Nelson [EMAIL PROTECTED] pgpRCgU4dRYAD.pgp Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?
Barton Fisher schrieb: It has IOS is 12.3(14)T7 - I'm wondering if this router can be made SIP aware. Apparently, this current firmware/programming is not, one way audio problems. Is there a version that support VoIP directly for this router? Do you have firewall feature set? Then you could simply activate the SIP protocol inspection. Without firewall feature set, I guess, it's impossible. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with MFC/R2
All channels 1~15, 17~31 is supposed to be double way. To place and receive calls. The line is supposed be E1-MFC/R2 and works perfect with a Panasonic PBX, actually Exists any variant of MFC/R2? And how can I configure it to get working? Your help will be very appreciated! Thank you! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Thursday, September 18, 2008 10:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Ok, in your E1 setup: 1-15: to outgoing calls 16-30: for incomming calls ? Now for make calls your telephone company must be provide MFC-R2 signaling. In your case the logs files show an invalid signal on make call. Regards, Luis Morales On Fri, Sep 19, 2008 at 10:10 AM, Dae Yeung Um [EMAIL PROTECTED] wrote: Hello I got: [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call called - 'g1/6055151' [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call caller id - '1102' [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: no UC_CATEGORY specified for chan UniCall/1-1, using default NATIONAL_SUBSCRIBER [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Call control(1) [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Make call [Sep 17 19:24:50] WARNING[4934] chan_unicall.c: Make call failed - Blocked [Sep 17 19:24:50] VERBOSE[4934] logger.c: -- Couldn't call g1/6055151 [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Channel gains [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Channel switching [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: Hangup: channel: 1 index = 0, normal = 11, callwait = -1, thirdcall = -1 [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: Updated conferencing on 1, with 0 conference users [Sep 17 19:24:50] VERBOSE[4934] logger.c: -- Hungup 'UniCall/1-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Thursday, September 18, 2008 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Dae, Activate debug full: asterisk -vr in other console do: tail -vf /var/log/asterisk/full Try to put call and send us more details about your logs Regards, Luis Morales On Thu, Sep 18, 2008 at 11:49 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: In fact I see 1101 in the rx bits on all channels... But I have in parallel one old Panasonic Key Phone system (Actually in production, to be replaced by asterisk), and it's works perfectly and immediately once I pass the E1 cables to there... So, the problem is not from Telco... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Wednesday, September 17, 2008 10:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 It seems to me your lines are blocked. Execute zttool and if you see 1101 in the rx bits, it means the telco (or whatever you have in the other end) has blocked their side. If this is a telco line you need to call them and tell them to unblock your lines. On Wed, Sep 17, 2008 at 10:33 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Thank you for the reply I shutdown asterisk and tried again and I have to following logs... OUTGOING TEST : Testcall.conf caller yes destination-no 6055151 originating-no 7309130 protocol-class mfcr2 protocol-variant ar,20,4 circuits 1-2 Log: ./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309130' to '6055151' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309131' to '6055152' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) MFC/R2 Chan 2: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Incoming test : Testcall.conf caller no protocol-class mfcr2 protocol-variant ar,20,4 on-offered answer circuits 1-2 Log: Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call
Re: [asterisk-users] device probe order question
On Thu, Sep 18, 2008 at 01:09:38PM -0400, Jason T. Nelson wrote: I have an Ubuntu 8 server running Asterisk 1.4 with three interface cards in it: * Wildcard TDM400P * Wildcard TDM410P * Wildcard TE122 I'm using zaptel 1.4.11, and the difficulty I'm running into is that with EVERY reboot, the order in which the hardware appears changes. This makes ztscan cough up a different order for the spans, which makes it nearly impossible to use the Digium GUI as after reboots, it complains that the hardware has changed. Is there a way to lock down probe order on boot, or somehow write zaptel.conf in such a way that port and span assignments never change between reboots? Those cards use each a different driver. Write those driver, in your preffered order, in /etc/modules . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with MFC/R2
I'm not sure but on E1 setup you can have only one way (in or out). In my case i have 15 in and 15 out. Told me more about your hardware: - E1 cards - How did you do to connect E1 interface to E1 asterisk's card ? - You can receive calls ? Please send us zapata.conf and unicall.conf Regards, Luis Morales On Fri, Sep 19, 2008 at 12:46 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: All channels 1~15, 17~31 is supposed to be double way. To place and receive calls. The line is supposed be E1-MFC/R2 and works perfect with a Panasonic PBX, actually Exists any variant of MFC/R2? And how can I configure it to get working? Your help will be very appreciated! Thank you! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Thursday, September 18, 2008 10:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Ok, in your E1 setup: 1-15: to outgoing calls 16-30: for incomming calls ? Now for make calls your telephone company must be provide MFC-R2 signaling. In your case the logs files show an invalid signal on make call. Regards, Luis Morales On Fri, Sep 19, 2008 at 10:10 AM, Dae Yeung Um [EMAIL PROTECTED] wrote: Hello I got: [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call called - 'g1/6055151' [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call caller id - '1102' [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: no UC_CATEGORY specified for chan UniCall/1-1, using default NATIONAL_SUBSCRIBER [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Call control(1) [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Make call [Sep 17 19:24:50] WARNING[4934] chan_unicall.c: Make call failed - Blocked [Sep 17 19:24:50] VERBOSE[4934] logger.c: -- Couldn't call g1/6055151 [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Channel gains [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Channel switching [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: Hangup: channel: 1 index = 0, normal = 11, callwait = -1, thirdcall = -1 [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: Updated conferencing on 1, with 0 conference users [Sep 17 19:24:50] VERBOSE[4934] logger.c: -- Hungup 'UniCall/1-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Thursday, September 18, 2008 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Dae, Activate debug full: asterisk -vr in other console do: tail -vf /var/log/asterisk/full Try to put call and send us more details about your logs Regards, Luis Morales On Thu, Sep 18, 2008 at 11:49 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: In fact I see 1101 in the rx bits on all channels... But I have in parallel one old Panasonic Key Phone system (Actually in production, to be replaced by asterisk), and it's works perfectly and immediately once I pass the E1 cables to there... So, the problem is not from Telco... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Wednesday, September 17, 2008 10:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 It seems to me your lines are blocked. Execute zttool and if you see 1101 in the rx bits, it means the telco (or whatever you have in the other end) has blocked their side. If this is a telco line you need to call them and tell them to unblock your lines. On Wed, Sep 17, 2008 at 10:33 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Thank you for the reply I shutdown asterisk and tried again and I have to following logs... OUTGOING TEST : Testcall.conf caller yes destination-no 6055151 originating-no 7309130 protocol-class mfcr2 protocol-variant ar,20,4 circuits 1-2 Log: ./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309130' to '6055151' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309131' to '6055152' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) MFC/R2 Chan 2: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Incoming test : Testcall.conf caller no protocol-class mfcr2 protocol-variant ar,20,4 on-offered
Re: [asterisk-users] device probe order question
In our last exciting episode, Tzafrir Cohen ([EMAIL PROTECTED]) said: Those cards use each a different driver. Write those driver, in your preffered order, in /etc/modules . Ah, I should have mentioned I did that (snippit from /etc/modules below) zaptel wcte12xp wctdm Having this doesn't seem to affect things. -- Jason T. Nelson [EMAIL PROTECTED] pgpjSTalNoGz0.pgp Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?
On Thu, 18 Sep 2008, Philipp Kempgen wrote: Gordon Henderson schrieb: If the web server is running php, then this will work: ? $action = $HTTP_GET_VARS[action] ; $file = $HTTP_GET_VARS[file] ; $caller = $HTTP_GET_VARS[caller] ; if (empty ($action) || empty ($file)) die (Something went wrong) ; // Open the file $fileName = /prefix/ . $file ; $fd = @fopen ($fileName, rb) ; Without any validation of the filename? It could be ../../secret/file. Left as an excercise to the user. That's not what I use 'for real', I just hacked out the relevant bits. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?
On Thu, Sep 18, 2008 at 12:20 PM, Barton Fisher [EMAIL PROTECTED] wrote: Hi, It has IOS is 12.3(14)T7 - I'm wondering if this router can be made SIP aware. Apparently, this current firmware/programming is not, one way audio problems. Is there a version that support VoIP directly for this router? Thanks, Bart Bart, IMNSHO, the less SIP aware the better... I have to disable SIP inspection on every IOS/PIX device I come across. Fix the one-way audio problems on your proxy, registrar, etc (in the case, Asterisk). Most SIP ALGs are broken. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to stream a A-Law/wav file to a browser ?
Olivier wrote: Hi, How can I create a web page allowing people to listen (with their own PC) a couple of .wav/a-law files stored on a Linux server ? Chances are users would access this web page from Internet Explorer but if I could make it available to other browsers, that would be better. I googled a bit and couldn't find a tag such as media://myaudiofile.wav that would fulfill this spec. As much as possible, I would be happy to avoid configuring browser plugins and so on. So if this media:// could be already installed and running in users PCs, that would be fine. I've read Red5 servers/Flash players combination could respond but I'm not too confident about a-law support and Red5 installation complexity for a 100% pure beginner. This works for IE. It displays the Play Button of the Windows Media Player. No need to download files or anything. It plays right off the web page: html body object classid=CLSID:6BF52A52-394A-11d3-B153-00C04F79FAA6 type=application/x-oleobject width=35 height=32 align=absmiddle id=VIDEO style=width: 35px; height: 30px; param name=URL value=http://your WAV url goes here param name=SendPlayStateChangeEvents value=True param name=AutoStart value=False param name=uiMode value=mini param name=PlayCount value=1 /object body /html Andres http://www.neuroredes.com What do you think ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium training course
On Thu, Sep 18, 2008 at 12:26 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Tilghman Lesher wrote: On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote: On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote: Anybody knows how to get a Coupon Code for the discount on the Asterisk training classes??? I am interested on taking that upcoming Asterisk Advance course, and 3K is kinda steep and considering I am still a college student paying this training out of my pocket, every bit helps. Sorry to thread jack. For that matter, I think old timers like myself should automatically get a dCAP. Six or seven years of Asterisk extensive experience should grandfather the dCAP and maybe even the training. I am sure I have a few tricks up my sleeve that the instructors don't know. If memory serves me correctly, there was talk about this very issue when the training and dCAP track came out. I will google it later. Nobody, including Mark Spencer and myself, have gotten a free pass on the dCAP. That said, I think you may be able to take the test for cheap. I don't know the exact price, but in any bootcamp, they allow people to come down on the last day and take only the test, as I did, if they have the space and resources (specifically, for the practical portion). I also used to think the old timers should get grandfathered dCAP cert. Then I looked at some of the stuff the dCAP certification tests for and realized that almost nobody out there that learned Asterisk from the docs and use it in a real world install would be able to pass the dCAP. As Tilghman said you can take JUST the dCAP test for a reasonable fee without having to take the classes. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. Another paper mill to bring down the reputation of the dCAP. Are there braindumps out there, or TroyTec (http://www.troytec.com) 13 page cheat papers that will allow you to hold the highly coveted dCAP? Maybe I should create a site for a nominal donation to the practice tests and braindumps. dCAP is useless if not based on real world experience. That is how I got my CCNA, real world experience. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] device probe order question
On Thu, Sep 18, 2008 at 01:50:42PM -0400, Jason T. Nelson wrote: In our last exciting episode, Tzafrir Cohen ([EMAIL PROTECTED]) said: Those cards use each a different driver. Write those driver, in your preffered order, in /etc/modules . Ah, I should have mentioned I did that (snippit from /etc/modules below) zaptel wcte12xp wctdm Having this doesn't seem to affect things. 'zaptel' is not really necessary . You don't have wctdm24xxp there. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium training course
Steve Totaro wrote: On Thu, Sep 18, 2008 at 12:26 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Tilghman Lesher wrote: On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote: On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote: Anybody knows how to get a Coupon Code for the discount on the Asterisk training classes??? I am interested on taking that upcoming Asterisk Advance course, and 3K is kinda steep and considering I am still a college student paying this training out of my pocket, every bit helps. Sorry to thread jack. For that matter, I think old timers like myself should automatically get a dCAP. Six or seven years of Asterisk extensive experience should grandfather the dCAP and maybe even the training. I am sure I have a few tricks up my sleeve that the instructors don't know. If memory serves me correctly, there was talk about this very issue when the training and dCAP track came out. I will google it later. Nobody, including Mark Spencer and myself, have gotten a free pass on the dCAP. That said, I think you may be able to take the test for cheap. I don't know the exact price, but in any bootcamp, they allow people to come down on the last day and take only the test, as I did, if they have the space and resources (specifically, for the practical portion). I also used to think the old timers should get grandfathered dCAP cert. Then I looked at some of the stuff the dCAP certification tests for and realized that almost nobody out there that learned Asterisk from the docs and use it in a real world install would be able to pass the dCAP. As Tilghman said you can take JUST the dCAP test for a reasonable fee without having to take the classes. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. Another paper mill to bring down the reputation of the dCAP. Are there braindumps out there, or TroyTec (http://www.troytec.com) 13 page cheat papers that will allow you to hold the highly coveted dCAP? Maybe I should create a site for a nominal donation to the practice tests and braindumps. dCAP is useless if not based on real world experience. That is how I got my CCNA, real world experience. If the end customer doesn't even know what asterisk is, what good is certification ? I get resumes all the time with certifications on them I have never even heard of - I just totally ignore that stuff and look at what they have actually been doing, if its been flipping burgers, it goes in the trash. Even if you can't get a job in the field you want, there is nothing stopping you from working on open source projects, trying stuff etc., and then put it on your resume. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with MFC/R2
The line is supposed be E1-MFC/R2 and works perfect with a Panasonic PBX, actually Exists any variant of MFC/R2? And how can I configure it to get working? As I said, no matter which variant you try, the AB bits MUST be in 10 to be able to make calls with Unicall/libmfcr2. I have never seen a variant which does not set bits AB in 10 for IDLE and 11 for BLOCKED. Most variants differ in the MF tones used, not on the R2 bits. Which telco is this and which country? you used Argentina, are you there? I am willing to troubleshoot your box if you give me access. You can contact me at google talk or msn at the same address you see in this e-mail. Moy -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Old voicemail bounces users
Folks, I have an odd problem (at least, it's odd to me). System language is spanish (es) and when users check their voicemail, if they don't delete it it goes into the Old directory. That's all well and good, but those users with messages in their Old directory try to get into voicemail and when the recording gets to you have 40 new and ... they hear the and and the system hangs them up. If I delete the messages in the Old directory, they are back in business. Until then, it's no go. I've gone over the messages in the Old directory and they are not damaged, permissions look good, so I'm not sure where the problem is. Any hints? TIA, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium training course
On Thu, 2008-09-18 at 14:22 -0400, Steve Totaro wrote: Are there braindumps out there, or TroyTec (http://www.troytec.com) 13 page cheat papers that will allow you to hold the highly coveted dCAP? Not that I'm aware of. dCAP is useless if not based on real world experience. That is how I got my CCNA, real world experience. One of the key tenets of the dCAP program is that it has both a written and practical portion of the test. For the sake of those that might not be familiar with the dCAP exam, let me explain: The dCAP exam has two portions... the first of which is a ninety-minute written exam. The written exam contains approximately 115 multiple-choice questions that range from channel driver configuration to dialplan applications to telephony concepts and everything in between. While it is something that you could theoretically cram for, we wrote many of the questions with real-world applications in mind, and tried to address items that you'd learn through real-world experience and not just book learning. The second portion of the exam is a ninety-minute practical exam. The idea of the practical exam is to treat you as if we'd hired you as an Asterisk consultant, and see if you can compile Asterisk from source and build the required PBX features and settings within the allotted time. Again, this is a test of your real-world Asterisk skills, and having proctored the exam for the past several years, I can state unequivocally that no amount of studying cheat sheets is going to prepare you for the practical exam like hands-on experience with Asterisk can. If there are any other questions I can answer about the dCAP exam (without giving away all the answers to the test, obviously), let me know and I'd be happy to address them. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Old voicemail bounces users
On Thursday 18 September 2008 14:19:29 David A. Bandel wrote: System language is spanish (es) and when users check their voicemail, if they don't delete it it goes into the Old directory. That's all well and good, but those users with messages in their Old directory try to get into voicemail and when the recording gets to you have 40 new and ... they hear the and and the system hangs them up. If I delete the messages in the Old directory, they are back in business. Until then, it's no go. I've gone over the messages in the Old directory and they are not damaged, permissions look good, so I'm not sure where the problem is. Are you missing vm-Old.gsm in your sounds directory? Please note that the name is case-sensitive on Linux and other Unix systems. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?
Kristian Kielhofner schrieb: IMNSHO, the less SIP aware the better... I have to disable SIP inspection on every IOS/PIX device I come across. Fix the one-way audio problems on your proxy, registrar, etc (in the case, Asterisk). Most SIP ALGs are broken. Interesting. I have my Asterisk with RFC-1918 IPs behid a NATting PIX and the FIXUP SIP of the PIX makes it very easy for me to use my * as server for external clients as well as as client for SIP providers. The PIX nicely replaces the RFC-1918 IP in the SIP-traffic with the current (dynamic) public IP of itself and keeps track of the RTP traffic. Actually, it also chages the ports in the RTP negotiation and then automatically forward the RTP traffic to the ports, the * was offering. Very very convenient. If the IOS firewall in the newer routers make problems, maybe I should not change to an ISR as I planned :). Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?
On Thu, Sep 18, 2008 at 4:18 PM, Stefan Gofferje [EMAIL PROTECTED] wrote: Interesting. I have my Asterisk with RFC-1918 IPs behid a NATting PIX and the FIXUP SIP of the PIX makes it very easy for me to use my * as server for external clients as well as as client for SIP providers. The PIX nicely replaces the RFC-1918 IP in the SIP-traffic with the current (dynamic) public IP of itself and keeps track of the RTP traffic. Actually, it also chages the ports in the RTP negotiation and then automatically forward the RTP traffic to the ports, the * was offering. Very very convenient. If the IOS firewall in the newer routers make problems, maybe I should not change to an ISR as I planned :). Terve, Stefan Stefan, Your version of PIX might have finally gotten it right, but even recent 12.4T IOS releases tend to really confuse NAT situations (same seems to go for various PIX releases I've used). Part of the problem might be the use of things like nathelper: http://www.iptel.org/ser/doc/modules/nathelper While not related to Asterisk, inconsistencies across SIP ALGs usually cause various ranges of flags passed to nat_uac_test to fail and/or turn up different results depending on what, specifically, the ALG is doing. NAT handling capabilities at the proxy/registrar, inconsistencies across SIP ALGs, dumb PATs not doing any specific protocol fixups (lowest common denominator), and the increasing use of SIP TLS (no ability to snoop/modify SIP headers or bodies including SDPs) tells me that SIP ALGs are not the best solution in most cases, certainly not long term. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom phones and DNS SRV
Just in case anyone is having DNS SRV timeouts with their Polycom phones, the following Polycom KB article should help: http://knowledgebase.polycom.com/kb/search.do?cmd=displayKCdocType=kcexternalId=12856sliceId=SAL_PUBLIC_1_2dialogID=7620671stateId=1 We have set tcpIpApp.port.rtp.mediaPortRangeStart to 65000. Based on our experience and the fact that the phone's DNS resolver starts over from port 1026 on a reboot and increments from there, this should give us about a year before the ports overlap again, in the unlikely event that the phones won't get rebooted in the meantime. YMMV. CP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] device probe order question
Hello, We had the same problem in the past and the last idea I had was to remove first the modules and load them (using /etc/rc.local) in the right order. Like: rmmod wcte11xp rmmod wctdm modprobe wcte11xp modprobe wctdm modprobe zaptel Maybe not the best way to do the job but it works for us. HTH, Ioan. On Thu, Sep 18, 2008 at 9:33 PM, Tzafrir Cohen [EMAIL PROTECTED]wrote: On Thu, Sep 18, 2008 at 01:50:42PM -0400, Jason T. Nelson wrote: In our last exciting episode, Tzafrir Cohen ([EMAIL PROTECTED]) said: Those cards use each a different driver. Write those driver, in your preffered order, in /etc/modules . Ah, I should have mentioned I did that (snippit from /etc/modules below) zaptel wcte12xp wctdm Having this doesn't seem to affect things. 'zaptel' is not really necessary . You don't have wctdm24xxp there. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Ioan (Nini) Indreias - [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get rid of Really destroying SIP dialog
2008/9/18 Tilghman Lesher [EMAIL PROTECTED] On Thursday 18 September 2008 05:16:21 Olivier wrote: Whatever the verbosity level (even 0), my Asterisk console is full of Really destroying SIP dialog messages. Is there a way to get rid of those ? Turn off debugging: core set debug 0 (and don't specify -d on your command line). So, I mixed up verbosity and debug. Thanks for your help. If not, do you think it deserves to marked as a bug ? It is a debugging message, not a bug. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium training course
On Thu, Sep 18, 2008 at 3:22 PM, Jared Smith [EMAIL PROTECTED] wrote: On Thu, 2008-09-18 at 14:22 -0400, Steve Totaro wrote: Are there braindumps out there, or TroyTec (http://www.troytec.com) 13 page cheat papers that will allow you to hold the highly coveted dCAP? Not that I'm aware of. dCAP is useless if not based on real world experience. That is how I got my CCNA, real world experience. One of the key tenets of the dCAP program is that it has both a written and practical portion of the test. For the sake of those that might not be familiar with the dCAP exam, let me explain: The dCAP exam has two portions... the first of which is a ninety-minute written exam. The written exam contains approximately 115 multiple-choice questions that range from channel driver configuration to dialplan applications to telephony concepts and everything in between. While it is something that you could theoretically cram for, we wrote many of the questions with real-world applications in mind, and tried to address items that you'd learn through real-world experience and not just book learning. The second portion of the exam is a ninety-minute practical exam. The idea of the practical exam is to treat you as if we'd hired you as an Asterisk consultant, and see if you can compile Asterisk from source and build the required PBX features and settings within the allotted time. Again, this is a test of your real-world Asterisk skills, and having proctored the exam for the past several years, I can state unequivocally that no amount of studying cheat sheets is going to prepare you for the practical exam like hands-on experience with Asterisk can. If there are any other questions I can answer about the dCAP exam (without giving away all the answers to the test, obviously), let me know and I'd be happy to address them. -- Jared Smith Training Manager Digium, Inc. Is google permitted, what OS, internet, can I install Lynx? ;-) Joking, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Voicemail emails
Depending on what e-mail server software you use, it may be easier to direct the voicemail to a specific e-mail address and have your e-mail software rewrite the subject, and then forward it on to your boss. On Thu, Sep 18, 2008 at 11:07 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Thursday 18 September 2008 09:54:39 Steve Anness wrote: So here is the deal. I have an Asterisk server here at work that I have recently taken over and the boss is wanting the server to do a lot of things that it didn't do before. I have already configured much of what he wanted including a voice messaging line where anyone can call in and leave a message and then he would get that message in his email. However, the boss wants his email subject to read something like This is an urgent message through the HISG voice messaging system so he knows that that message came through that number as opposed to his voicemail box that already gets forwarded there. The default is the [PBX]: New Message 10 in mailbox 0307. At second glance he would know which voicemail box is his line but he wants things to be different and so I am trying to make that happen. I know there is the 'emailsubject' option. I haven't tried this yet but my concern is that it will set the subject the same on every single box (obviously what the command is designed for). I can I customize a voicemail message so that if something comes in on our 0307 line it has a certain message and then we might get a message on 1942 line that we want a different subject. Currently, there is no such capability. Coding it would not be very difficult, however. I would suggest using the per-mailbox settings and simply adding the option to code an email subject per mailbox, and then default to the generic subject, if one is not otherwise specified in the mailbox. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Weavver. Your voice, just better. Business Development: +1.714.726.8071 XMPP: mitchel.at.weavver.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] device probe order question
That is how I do it as well but don't forget /usr/sbin/asterisk or you will just have a bunch of loaded modules. I never bother with init scripts or /etc/modules. rc.local all the way and I once challenged the list to give me a reason why that is NOT a good way. No replies... Thanks, Steve Totaro On Thu, Sep 18, 2008 at 5:24 PM, Ioan Indreias [EMAIL PROTECTED] wrote: Hello, We had the same problem in the past and the last idea I had was to remove first the modules and load them (using /etc/rc.local) in the right order. Like: rmmod wcte11xp rmmod wctdm modprobe wcte11xp modprobe wctdm modprobe zaptel Maybe not the best way to do the job but it works for us. HTH, Ioan. On Thu, Sep 18, 2008 at 9:33 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Sep 18, 2008 at 01:50:42PM -0400, Jason T. Nelson wrote: In our last exciting episode, Tzafrir Cohen ([EMAIL PROTECTED]) said: Those cards use each a different driver. Write those driver, in your preffered order, in /etc/modules . Ah, I should have mentioned I did that (snippit from /etc/modules below) zaptel wcte12xp wctdm Having this doesn't seem to affect things. 'zaptel' is not really necessary . You don't have wctdm24xxp there. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Ioan (Nini) Indreias - [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Voicemail emails
We have done something similar using the category option with the voicemail. Our emails look like this: -- TO : Big Boss ID : 2 CAT. : EMERGENCY BOX : 100 FROM : Emergency Line 5552221212 DUR : 0:20 DATE : Wednesday, October 10, 2007 at 01:28:27 PM -- Internal Access: *98 for Personal Voicemail or *99 for Main Voicemail You could easily append this to the subject line, so it will show different per category. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mitcheloc Sent: Thursday, September 18, 2008 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Custom Voicemail emails Depending on what e-mail server software you use, it may be easier to direct the voicemail to a specific e-mail address and have your e-mail software rewrite the subject, and then forward it on to your boss. On Thu, Sep 18, 2008 at 11:07 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Thursday 18 September 2008 09:54:39 Steve Anness wrote: So here is the deal. I have an Asterisk server here at work that I have recently taken over and the boss is wanting the server to do a lot of things that it didn't do before. I have already configured much of what he wanted including a voice messaging line where anyone can call in and leave a message and then he would get that message in his email. However, the boss wants his email subject to read something like This is an urgent message through the HISG voice messaging system so he knows that that message came through that number as opposed to his voicemail box that already gets forwarded there. The default is the [PBX]: New Message 10 in mailbox 0307. At second glance he would know which voicemail box is his line but he wants things to be different and so I am trying to make that happen. I know there is the 'emailsubject' option. I haven't tried this yet but my concern is that it will set the subject the same on every single box (obviously what the command is designed for). I can I customize a voicemail message so that if something comes in on our 0307 line it has a certain message and then we might get a message on 1942 line that we want a different subject. Currently, there is no such capability. Coding it would not be very difficult, however. I would suggest using the per-mailbox settings and simply adding the option to code an email subject per mailbox, and then default to the generic subject, if one is not otherwise specified in the mailbox. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Weavver. Your voice, just better. Business Development: +1.714.726.8071 XMPP: mitchel.at.weavver.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Streaming MoH on 1.4
- Olivier [EMAIL PROTECTED] wrote: A somehow related question, is broadcasting streaming music as music on hold, submitted to any licencing fee ? I got here late. The only way you can legally use music as music on hold is if you either pay, or are not subject to pay, performance royalty money to *someone*. Who you might pay includes BMI, ASCAP, and SESAC, who have standardized annual blanket licenses for that sort of thing, which permit you to play any music to which they've been assigned the right to collect and disburse such monies. Or, if you have recordings directly from an act who have not sold their rights to, say, a music label, they could license you directly. Or you could play the music yourself. But note that if you do *that*, while you aren't liable for performance royalties, you as a performer will own the songwriter(s) money, usually in the form of compulsory mechanical royalties. How those are handled if you record your own arrangement of Hey Jude once and loop it on music on hold, I'm not clear on. No, IANAL. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] device probe order question
Steve Totaro schrieb: I never bother with init scripts or /etc/modules. rc.local all the way and I once challenged the list to give me a reason why that is NOT a good way. No replies... http://lists.digium.com/pipermail/asterisk-users/2008-April/210491.html http://lists.digium.com/pipermail/asterisk-users/2008-April/210498.html http://lists.digium.com/pipermail/asterisk-users/2008-May/212566.html etc. :-) Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?
I've had the same experience. I probably have 20-30 customers with multiple SIP phones behind PIX running 6.3(5) (which has been out almost 3 years) and I have no issues at all. You can even have two phones behind a PIX being PAT'd to a single external IP with reinvite enabled in * and you will still get 2 way audio. The SIP Fixup makes changes inside the SIP packet for internal IPs. The nice thing is that you don't need to enable NAT on the remote * server either. It thinks the device is not behind NAT. I have customers with 20 phones behind one IP connecting to a remote * box with no issues at all and no special PIX config. Now the IOS firewall, that is a completely different animal and works completely different than the PIX/ASA. Stefan Gofferje wrote: Kristian Kielhofner schrieb: IMNSHO, the less SIP aware the better... I have to disable SIP inspection on every IOS/PIX device I come across. Fix the one-way audio problems on your proxy, registrar, etc (in the case, Asterisk). Most SIP ALGs are broken. Interesting. I have my Asterisk with RFC-1918 IPs behid a NATting PIX and the FIXUP SIP of the PIX makes it very easy for me to use my * as server for external clients as well as as client for SIP providers. The PIX nicely replaces the RFC-1918 IP in the SIP-traffic with the current (dynamic) public IP of itself and keeps track of the RTP traffic. Actually, it also chages the ports in the RTP negotiation and then automatically forward the RTP traffic to the ports, the * was offering. Very very convenient. If the IOS firewall in the newer routers make problems, maybe I should not change to an ISR as I planned :). Terve, Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] device probe order question
Philipp Kempgen wrote: Steve Totaro schrieb: I never bother with init scripts or /etc/modules. rc.local all the way and I once challenged the list to give me a reason why that is NOT a good way. No replies... http://lists.digium.com/pipermail/asterisk-users/2008-April/210491.html http://lists.digium.com/pipermail/asterisk-users/2008-April/210498.html http://lists.digium.com/pipermail/asterisk-users/2008-May/212566.html etc. :-) Philipp Kempgen Also remeber that using init scripts gives the posibility of running those same scripts to stop the services in reverse order when you shutdown a machine. If you simply start programs in rc.local, when you shutdown the machine it won't stop your programs elegantly. (for example, just look inside /etc/rc3.d and you will see all the start scripts 'S' which run when you bootup the machine and the kill scripts 'K' which run when you shutdown the machine). Andres, http://www.neuroredes.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with MFC/R2
It's a Digium TE121P with Echo Cancellation Zapata.conf # Span 1: WCT1/0 Wildcard TE121 Card 0 HDB3/CCS/CRC4 RED RECOVERING span=1,1,0,ccs,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 # Span 2: WCTDM/0 Wildcard AEX800 Board 1 (MASTER) fxsks=32 fxsks=33 fxsks=34 fxsks=35 # channel 36, WCTDM, no module. # channel 37, WCTDM, no module. # channel 38, WCTDM, no module. # channel 39, WCTDM, no module. # Global data loadzone= us defaultzone = us [Channels] language=en usecallerid=yes echocancel=yes rxgain=0 txgain=0 group=1 callgroup=0 pickupgroup=0 amaflags=default accountcode=avantel musiconhold=default context=from-pstn group=1 loglevel=0 protocolclass=mfcr2 protocolvariant=ar,20,4 channel = 1-15 channel = 16-31 I cannot receive calls... I cant see any type of logs on the console when I try to call in. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Thursday, September 18, 2008 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 I'm not sure but on E1 setup you can have only one way (in or out). In my case i have 15 in and 15 out. Told me more about your hardware: - E1 cards - How did you do to connect E1 interface to E1 asterisk's card ? - You can receive calls ? Please send us zapata.conf and unicall.conf Regards, Luis Morales On Fri, Sep 19, 2008 at 12:46 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: All channels 1~15, 17~31 is supposed to be double way. To place and receive calls. The line is supposed be E1-MFC/R2 and works perfect with a Panasonic PBX, actually Exists any variant of MFC/R2? And how can I configure it to get working? Your help will be very appreciated! Thank you! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Thursday, September 18, 2008 10:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Ok, in your E1 setup: 1-15: to outgoing calls 16-30: for incomming calls ? Now for make calls your telephone company must be provide MFC-R2 signaling. In your case the logs files show an invalid signal on make call. Regards, Luis Morales On Fri, Sep 19, 2008 at 10:10 AM, Dae Yeung Um [EMAIL PROTECTED] wrote: Hello I got: [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call called - 'g1/6055151' [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call caller id - '1102' [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: no UC_CATEGORY specified for chan UniCall/1-1, using default NATIONAL_SUBSCRIBER [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Call control(1) [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Make call [Sep 17 19:24:50] WARNING[4934] chan_unicall.c: Make call failed - Blocked [Sep 17 19:24:50] VERBOSE[4934] logger.c: -- Couldn't call g1/6055151 [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Channel gains [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Channel switching [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: Hangup: channel: 1 index = 0, normal = 11, callwait = -1, thirdcall = -1 [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: Updated conferencing on 1, with 0 conference users [Sep 17 19:24:50] VERBOSE[4934] logger.c: -- Hungup 'UniCall/1-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Thursday, September 18, 2008 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Dae, Activate debug full: asterisk -vr in other console do: tail -vf /var/log/asterisk/full Try to put call and send us more details about your logs Regards, Luis Morales On Thu, Sep 18, 2008 at 11:49 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: In fact I see 1101 in the rx bits on all channels... But I have in parallel one old Panasonic Key Phone system (Actually in production, to be replaced by asterisk), and it's works perfectly and immediately once I pass the E1 cables to there... So, the problem is not from Telco... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Wednesday, September 17, 2008 10:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 It seems to me your lines are blocked. Execute zttool and if you see 1101 in the rx bits, it means the telco (or whatever you have in the other end) has blocked their side. If this is a telco line you need to call them and tell them to unblock your lines. On Wed, Sep 17, 2008 at 10:33 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Thank you for the reply I shutdown asterisk and tried again and I have to following logs... OUTGOING TEST : Testcall.conf
Re: [asterisk-users] Help with MFC/R2
Hi Dae, In zaptel.conf change ccs for cas and comment dchan line, for example: span=1,1,0,cas,hdb3 cas=1-15:1101 #dchan=16 cas=17-31:1101 -- Humberto Figuera - Using Linux 2.6.22 Usuario GNU/Linux 369709 Caracas - Venezuela GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium training course
I got my dCAP by turning up to the exam at Astricon in Madrid a couple years ago without doing any training. It may have changed since then but I found that the practical exam would be difficult if not impossible to pass without knowing what you were doing - either through real world experience or having done the training. I felt at the time the written portion was heavily biased towards people who had done the training - in fact I would go so far as to say that it was designed specifically to discriminate against people who had not attended the official training. Anyhow, the point I am making is that a brain dump will help you pass the written but you'll be humiliated (and rightly so) when you sit the practical. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 19 September 2008 2:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium training course On Thu, Sep 18, 2008 at 12:26 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Tilghman Lesher wrote: On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote: On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote: Anybody knows how to get a Coupon Code for the discount on the Asterisk training classes??? I am interested on taking that upcoming Asterisk Advance course, and 3K is kinda steep and considering I am still a college student paying this training out of my pocket, every bit helps. Sorry to thread jack. For that matter, I think old timers like myself should automatically get a dCAP. Six or seven years of Asterisk extensive experience should grandfather the dCAP and maybe even the training. I am sure I have a few tricks up my sleeve that the instructors don't know. If memory serves me correctly, there was talk about this very issue when the training and dCAP track came out. I will google it later. Nobody, including Mark Spencer and myself, have gotten a free pass on the dCAP. That said, I think you may be able to take the test for cheap. I don't know the exact price, but in any bootcamp, they allow people to come down on the last day and take only the test, as I did, if they have the space and resources (specifically, for the practical portion). I also used to think the old timers should get grandfathered dCAP cert. Then I looked at some of the stuff the dCAP certification tests for and realized that almost nobody out there that learned Asterisk from the docs and use it in a real world install would be able to pass the dCAP. As Tilghman said you can take JUST the dCAP test for a reasonable fee without having to take the classes. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. Another paper mill to bring down the reputation of the dCAP. Are there braindumps out there, or TroyTec (http://www.troytec.com) 13 page cheat papers that will allow you to hold the highly coveted dCAP? Maybe I should create a site for a nominal donation to the practice tests and braindumps. dCAP is useless if not based on real world experience. That is how I got my CCNA, real world experience. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Follow Me app question
Hi all, When one uses the follow-me logic to forward calls to lots of phone devices do subsequent calls get routed to the VM (or whatever the 10x is)? In other words, if I want my office, house and cell phones to ring whenever a call comes in and I answer it on my cell, does the next call that comes in when I'm on my cell get sent to VM or does it ring the follow-me group again? -- Mark Phillips, G7LTT/NI2O Randolph, NJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Old voicemail bounces users
On Thu, Sep 18, 2008 at 2:52 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Thursday 18 September 2008 14:19:29 David A. Bandel wrote: System language is spanish (es) and when users check their voicemail, if they don't delete it it goes into the Old directory. That's all well and good, but those users with messages in their Old directory try to get into voicemail and when the recording gets to you have 40 new and ... they hear the and and the system hangs them up. If I delete the messages in the Old directory, they are back in business. Until then, it's no go. I've gone over the messages in the Old directory and they are not damaged, permissions look good, so I'm not sure where the problem is. Are you missing vm-Old.gsm in your sounds directory? Please note that the name is case-sensitive on Linux and other Unix systems. [EMAIL PROTECTED]:~# find /var/lib/asterisk -name vm-Old.gsm -ls 32059204 -rw-r--r-- 1 asterisk asterisk 1518 Mar 5 2008 /var/lib/asterisk/sounds/es/vm-Old.gsm 32054554 -rw-r--r-- 1 asterisk asterisk 1023 Mar 5 2008 /var/lib/asterisk/sounds/vm-Old.gsm Not missing. vm-Old also exists as other codecs (ulaw and alaw). Ciao, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] what codec is sip using?
If you use iax, the console will tell you what codec is being used. But for sip, nothing is shown. With sip debug on, I get: Capabilities: us - 0x130e (gsm|ulaw|alaw|g729|speex|g722), peer - audio=0x100e (gsm|ulaw|alaw|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100e (gsm|ulaw|alaw|g722) but I don't see anything that shows which codec was used. How do I find out? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what codec is sip using?
Sean, Try 'sip show channels' or 'sip show channel channelid' for the drill down. I believe the codec in use will be displayed with either command. Dave From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of sean darcy [EMAIL PROTECTED] Sent: Thursday, September 18, 2008 10:49 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] what codec is sip using? If you use iax, the console will tell you what codec is being used. But for sip, nothing is shown. With sip debug on, I get: Capabilities: us - 0x130e (gsm|ulaw|alaw|g729|speex|g722), peer - audio=0x100e (gsm|ulaw|alaw|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100e (gsm|ulaw|alaw|g722) but I don't see anything that shows which codec was used. How do I find out? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk REFER
is this a feature in asterisk? On Mon, Sep 15, 2008 at 3:20 AM, Patrick Maartense [EMAIL PROTECTED]wrote: Ice is the feature you're looking for I think If two clients support ice, a direct link between them will be made -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Al lists *Sent:* Dienstag, 09. September 2008 23:40 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Asterisk REFER Hi All, from what i'm understanding, Asterisk is back to back user agent. Base on this my initial thought was even if we enable reinvite in sip.conf, asterisk still will be in sip path after transfer. But i read some information in asterisk using refer to transfer a call completely to another sip or per say, a call comes in from voip provider and get transferred by asterisk to a cell phone number by using same provider and then asterisk will not be in SIP path anymore. is it doable ? No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.169 / Virus Database: 270.6.19/1661 - Release Date: 09.09.2008 04:58 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what codec is sip using?
David Gibbons wrote: Sean, Try 'sip show channels' or 'sip show channel channelid' for the drill down. I believe the codec in use will be displayed with either command. Dave Thanks that worked. Now how do I get it show the codec when I'm not at the CLI? sean From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of sean darcy [EMAIL PROTECTED] Sent: Thursday, September 18, 2008 10:49 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] what codec is sip using? If you use iax, the console will tell you what codec is being used. But for sip, nothing is shown. With sip debug on, I get: Capabilities: us - 0x130e (gsm|ulaw|alaw|g729|speex|g722), peer - audio=0x100e (gsm|ulaw|alaw|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100e (gsm|ulaw|alaw|g722) but I don't see anything that shows which codec was used. How do I find out? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what codec is sip using?
sean darcy wrote: David Gibbons wrote: Sean, Try 'sip show channels' or 'sip show channel channelid' for the drill down. I believe the codec in use will be displayed with either command. Dave Thanks that worked. Now how do I get it show the codec when I'm not at the CLI? Show it where, if not the CLI? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T100P detection.
Greetings, I am running kernel 2.6.26.5, Asterisk 1.6.0rc2 and DAHDI 2.0.0rc4/rc2 and cannot get the DAHDI drivers to detect my Digium T100P: 01:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface The message when loading the wct1xxp module is: t1xxp: probe of :01:00.0 failed with error -5 As a result, I cannot use the 1.6 beta with this card. It works fine with Zaptel 1.4.12.1 and Asterisk 1.4.21.1: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.12.1 Zaptel Echo Canceller: MG2 Registered Tormenta2 PCI Framer: DS21552, Revision: 3 (T1) Found a Wildcard: Digium Wildcard T100P T1/PRI Has anyone run into this? Does anyone know what the deal is with that? I'd dearly prefer to use 1.6. Cheers, -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line
On Wed, Sep 17, 2008 at 1:10 PM, logan [EMAIL PROTECTED] wrote: Thanks a lot Nhadie. I appreciate your help. Could you also suggest some brands or models of the FXO+FXS card that are seamlessly compatible to Asterisk? Also what hardphone I should go for as there are so many in the market? What should be the configuration of the system running this kind of Asterisk setup? And which Linux distribution is best suited with Asterisk? Hi you can look this compatable hardware http://www.voip-info.org/wiki/ http://www.voip-info.org/wiki/view/PSTN+Interface+Hardware+for+Computer+Systems http://www.voip-info.org/wiki/view/VOIP+Phones Its very difficult to say which OS is good, its all depends on your experience and your hands on the same. Look at Trixbox, its automated CD ram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
