Re: [asterisk-users] How to create or test tone configuration to include them in zaptel
On 7/16/06, Erick Perez [EMAIL PROTECTED] wrote: Hi, I would like to know what kind of tests should I make in order to document tone/configuration settings for analog cards and E1 cards specifically for my country (Panama). For example: Australia, Venezuela, etc, they have been documented and included in the zaptel config. Thanks, Hi You can use ztmonitor from the zaptel source to record the channel data. Later on use wavesurfer to analyze the tones. more at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+indications.conf Note that indications and zonedata.c use might share the same information but perform different functions: zonedata.c is for the zaptel hardware incoming calls while indications is used by asterisk it self with the Playtones application. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cheers, Maxim Vexler Free as in Freedom - Do u GNU ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SRTP enabling
Hi everyone, I was trying to support SRTP in asterisk for our Linksys IP Phones to prevent of ISP blocking issue. I compiled successfully SRTP from http://srtp.sourceforge.net/srtp.html But i don't know from where i should start to configure in Asterisk. Could someone please give me the example sip.conf for the way how i can support? You replies will be high appriciated. Abdul __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + fax
Maxim Vexler [EMAIL PROTECTED] wrote:On 7/12/06, al gav <[EMAIL PROTECTED]>wrote: Hi all I need a help with asterisk+fax - fax to email I am trying to setup fax to email with asterisk with no success. I have asterisk 1.2.9.1 running on CentOS i have created extension 300 which should receive faxes. extensions.conf - exten = 300,1,Goto(fax,s,1) exten = 300,2,Congestion exten = 300,3,Hangup exten = s,1,Macro(faxreceive) exten = h,1,system(/usr/bin/mail -s "Fax from ${CALLERIDNUM} ${CALLERIDNAME}" ${EMAILADDR} ${FAXFILE}) [macro-faxreceive] exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif) exten = s,2,Set(EMAILADDR=${FAX_RX_EMAIL}) exten = s,3,rxfax(${FAXFILE}|debug) exten = s,103,Set(EMAILADDR=${FAX_RX_EMAIL}) exten = s,104,Goto(3) When i am trying to call 300 extension i am receiving broken fax noise. in addition on the CLI i see the next line Executing RxFAX("SIP/5060-08d6f170", "/var/spool/asterisk/fax/1152714504.466.tif|debug") in new stack But the file never been created. In /var/log/asterisk/full i see these lines: Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up Jul 12 17:28:24 DEBUG[25029]
app_rxfax.c: FLOW HDLC carrier down Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up Any help with fax to email with Asterix will be appreciated. Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.
___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersWell, the fact that you are calling the extension and hear the faxsignaling shows the that rxfax application is working, so far so good.Have you set faxdetect=incoming in /etc/asterisk/zapata.conf ?-- Yes -- Also note that you need to have an extension named "fax" in your"default" context. In order to check the fax i put directly extension in the default context [default] exten =
300,1,Macro(faxreceive) Also look at http://www.voip-info.org/wiki-Asterisk+fax for helping info. It's sounds like the difference between the speeds of the faxes on bothsides, but i am not sure about that.Again the goal is to receive fax on asterisk and mail them to different mails.Thank you Maxim. -- Cheers,Maxim Vexler"Free as in Freedom" - Do u GNU ?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options
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Re: [asterisk-users] Asterisk instances on VPS
Salve James, *! On Sun, 16 Jul 2006, James Sturges wrote: Don't know if it helps, but in AU you can tell the telco to place all calls on 2 ISDN's at the same time. That way you could have 2 ISDN lines on 2 ISDN cards (or Spans) and all calls would be presented on both ISDN services. I thought that S2M (ISDN multiplex connection) does have a bus like S0 after th terminal adapter... With 2B+2 and S0 you connect up to 8 devices, so you don't need a second ISDN line - but what's about S2M? Isn't S2M a bus where you can connect more then one device? rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 300 headset with static noise
On Jul 14, 2006, at 10:13 PM, Adrià Vidal wrote: Someone using these phone Snom 300 with his own headset ? We got horrible static noise on them? P.D. Got silence as answer from Snom by now... maybe on holidays or with in the European Football championship. Have a look at this document: http://www.snom.com/wiki/index.php/ FAQs#Q:_Why_is_there_a_humming_noise_when_using_the_headset.3F Michiel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk instances on VPS
Hi there, Don't know if it helps, but in AU you can tell the telco to place all calls on 2 ISDN's at the same time. Same in Germany at Telekom: Standard BRI (2B+D) can be grouped together onto the same number. But, I know just applications of this with the point-to-point form of the protocol, not point-to-multipoint ... That way you could have 2 ISDN lines on 2 ISDN cards (or Spans) and all calls would be presented on both ISDN services. I thought that S2M (ISDN multiplex connection) does have a bus like S0 after th terminal adapter... With 2B+2 and S0 you connect up to 8 devices, so you don't need a second ISDN line - But beware, you can not use more than 2 devices concurrently ... but what's about S2M? Isn't S2M a bus where you can connect more then one device? S2M is point-to-point, no bus structure ... 30 BRI channels are packed into one S2M packet and must be unpacked by the PRI-adapter ... Jürgen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Snom 300 headset with static noise
On Freitag, 14. Juli 2006 10:13 Adrià Vidal wrote: Someone using these phone Snom 300 with his own headset ? We used to but the quality was horrifying. Since we changed to Plantronics Noise Cancelling headsets everything is wounderful. We got horrible static noise on them? Maybe the article Michiel pointed out helps you still the overall voice quality of their headsets (at least the ones they sold last year) is awful. Kind regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 7970 SIP configs
Hi All Has anyone got an annotated SEPmac.cnf.xml they are using successfully with the 7970 (8.0.3 Sip) and Asterisk? The SEPmac.cnf.xml files on the wiki are not annotated and although I've managed to upgrade the phone firmware and get a partial registration better info could speed it up. Is there a separate 7970 SIP forum/list anywhere? Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DUNDI / regcontext
Could you possibly put up the relevant section(s) of your sip.conf? It sounds like the DUNDi portion is set up properly, and obviously it's not going to find an extension that doesn't exist. Regards, - Brad From: [EMAIL PROTECTED] on behalf of Simon Woodhead Sent: Sat 7/15/2006 5:59 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DUNDI / regcontext Hi folks, I've been having a go at getting DUNDI working this evening to enable users to register to any Asterisk box and to look them up from another. The DUNDI part works just great (very impressed), as does the subsequent joining of calls between the two servers but I'm struggling with regcontext and would be grateful for any input. sip.conf includes: [general] regcontext=sipregistration When a user registers, I get the Added extension 'XX' priority 1 to sipregistration message. However, 'show dialplan' does not show the extension and a DUNDI lookup does not return it. The sipregistration context has been auto-created but is empty. If I manually create the sipregistration context and add the NoOp extension, then everything works as expected. I've tried this across multiple boxes, each running different versions right up to the latest stable but the behaviour is the same. It is also the same with both SIP and IAX registrations and doesn't make a difference if the peer is defined in the .conf file or Realtime. They do all have identical configurations though so I suspect there might be something in our setup which is conflicting. Any input gratefully received. All the best, Simon The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SRTP enabling
Abdul Lateef wrote: Hi everyone, I was trying to support SRTP in asterisk for our Linksys IP Phones to prevent of ISP blocking issue. I compiled successfully SRTP from http://srtp.sourceforge.net/srtp.html But i don't know from where i should start to configure in Asterisk. Could someone please give me the example sip.conf for the way how i can support? You replies will be high appriciated. Most of the blocking in other countries, was not for RTP traffic, but for signaling traffic (SIP usually, Mexico x Vonage comes to mind). You are sure they are blocking RTP traffic ? And, from what I understand, in some places the gov. forced the ISPs to remove the blocking (at least, I heard of one such a case in Brazil, a DSL provider started to block SIP, and Anatel, Brazil gov. entity that regulate telephony and others, asked them to remove the blocking, others with more knowledge of the case may be able to add their remarks) Blocking SIP if you control the server is somewhat easy to prevent (if is a plain dumb UDP port 5060 filtering), just have your server listen in another UDP port... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Skype protocol cracked?
http://news.yahoo.com/s/zd/20060714/tc_zd/183411 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sending flash using DTMF
Is there is a way to send Asterisk FLASH using DTMF? I am trying to redial or dialing a new number without hangingup and start the whole process again. Thanks, Osama ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending flash using DTMF
Yes just use features.conf On 7/16/06, Osama Kamal [EMAIL PROTECTED] wrote: Is there is a way to send Asterisk FLASH using DTMF? I am trying to redial or dialing a new number without hangingup and start the whole process again. Thanks, Osama ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Injecting prerecorded audio into active call
Wondering if someone else has ever done anything like this, or has any ideas if it is in fact possible. We currently record all our calls which are stored in gsm format. They are not recorded by asterisk, rather a 3rd party system, but we would be looking at using asterisk to implement this idea. We would like to train staff in remote offices over the phone. And at times we would like to play back a recorded call to the remote staff member during the call. We would look to do this via a web interface. So while the call is active, the trainer could login to the webinterface, somehow select the active call to use, then pick a call from a list and play it on the active call. Now I'm sure this is possible, but the question is how hard... If anyone has any ideas on this I would greatly appreciate it. Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Wrong account code from iax_buddies
Trixbox is only used as the client to simulate what we already saw happening with a customer. I don't think the fact that we used a Trixbox on the client side has anything to do with the problem on the server side which is not using Trixbox. The on the server side Asterisk only sees the Trixbox as a client coming in under an iax_buddies record which has an accountcode assigned to it. The server side is taking the call, then inserting the CDR with a correct account code, dst, src and so on but the channel is wrong. On 7/15/06, Tim Panton [EMAIL PROTECTED] wrote: On 15 Jul 2006, at 19:26, voiplist wrote: On 7/15/06, Tim Panton [EMAIL PROTECTED] wrote: On 15 Jul 2006, at 17:24, voiplist wrote: After further testing, here is what we found.. The account code was actually right after all, what made us think it was incorrect was the fact that * was reporting the wrong channel for the call. Test 2: We turned off the softphones mentioned above. We then setup a test Asterisk box with the same IAX accounts mentioned above. Each account registered to our server remotely just as the softphones did. We then made 10 phone calls with each account dialing totally different phone numbers so we could identify the records from each IAX account. Results (Test 2): We found that the majority of calls now had wrong/mismatched channels in the CDR. Basically we would show a channel like IAX2/user2 for a call that we are certain came from the IAX user user1. And... That's my story and I'm stickin' to it :) Hmm, I'd like to know more about this, as it is likely to bite me too :-( Can you describe the setup of the 'originating' asterisk in your second test? If possible please share the relevant bits of extensions.conf and iax.conf. Thanks. Tim. Tim Panton [EMAIL PROTECTED] Sure it was TrixBox 1.1 running on a VmWare virtual machine (great for testing). We basically setup two Trunks as they are known is TrixBox ([EMAIL PROTECTED]), one with each of the IAX accounts we setup on the server end. Just to keep the confusion to a minimum, the TrixBox was the client which was sending the calls to the Asterisk Server which was terminating the calls to the PSTN. If you are familiar with TrixBox Trunk Configs I can send you the basic details of the settings in the GUI Trunk screen. Ah, no, I am using the base asterisk facilities, plus a bit of AGI magic. You would need to strip the test case down to just asterisk before you could be sure where the problem is. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Injecting prerecorded audio into active call
in my knowledge the only interaction with asterisk audio channels is through eagi (refer to http://www.voip-info.org/wiki-Asterisk+AGI ) but as you can see there is no way to inject/add/mix audio please tell me i'm wrong On Mon, 2006-07-17 at 00:54 +0800, [EMAIL PROTECTED] wrote: Wondering if someone else has ever done anything like this, or has any ideas if it is in fact possible. We currently record all our calls which are stored in gsm format. They are not recorded by asterisk, rather a 3rd party system, but we would be looking at using asterisk to implement this idea. We would like to train staff in remote offices over the phone. And at times we would like to play back a recorded call to the remote staff member during the call. We would look to do this via a web interface. So while the call is active, the trainer could login to the webinterface, somehow select the active call to use, then pick a call from a list and play it on the active call. Now I'm sure this is possible, but the question is how hard... If anyone has any ideas on this I would greatly appreciate it. Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?
Hello list I'm trying to setup asterisk as an answering machine. How can I set asterisk to Answer() incoming call ONLY after specified count of ring cycles ? In the current situation I have the PBX connected to a home line, where POTS device are also connected on the same circuit. What I'm trying to do is allow a grace period where a POTS device could be picked up and those stop the ring indication on the line by this causing asterisk to not answer the call. In present situation even if the incoming phone call is taken off hook by a POST device asterisk still starts playing its incoming call IVR after the specified(where?) number of seconds. Thank you. -- Cheers, Maxim Vexler Free as in Freedom - Do u GNU ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Injecting prerecorded audio into active call
Well, if the web interface copied the call to a standard name and you had an extension using Playback or ControlPlayback to play that file and then bridged the call -- maybe that wold work -- much of a kludge though. on Sunday 07/16/2006 mike([EMAIL PROTECTED]) wrote in my knowledge the only interaction with asterisk audio channels is through eagi (refer to http://www.voip-info.org/wiki-Asterisk+AGI ) but as you can see there is no way to inject/add/mix audio please tell me i'm wrong On Mon, 2006-07-17 at 00:54 +0800, [EMAIL PROTECTED] wrote: Wondering if someone else has ever done anything like this, or has any ideas if it is in fact possible. We currently record all our calls which are stored in gsm format. They are not recorded by asterisk, rather a 3rd party system, but we would be looking at using asterisk to implement this idea. We would like to train staff in remote offices over the phone. And at times we would like to play back a recorded call to the remote staff member during the call. We would look to do this via a web interface. So while the call is active, the trainer could login to the webinterface, somehow select the active call to use, then pick a call from a list and play it on the active call. Now I'm sure this is possible, but the question is how hard... If anyone has any ideas on this I would greatly appreciate it. Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?
On Jul 16, 2006, at 11:36 AM, Maxim Vexler wrote: Hello list I'm trying to setup asterisk as an answering machine. How can I set asterisk to Answer() incoming call ONLY after specified count of ring cycles ? In the current situation I have the PBX connected to a home line, where POTS device are also connected on the same circuit. What I'm trying to do is allow a grace period where a POTS device could be picked up and those stop the ring indication on the line by this causing asterisk to not answer the call. In present situation even if the incoming phone call is taken off hook by a POST device asterisk still starts playing its incoming call IVR after the specified(where?) number of seconds. I don't think you can do that, since asterisk has no way to know when the shared PSTN line is answered by your analog phones... I don't think asterisk counts the rings, as much as it waits for answered status, which it is never going to see in your current configuration. I am a relative newb though, so maybe someone else here has a brilliant idea for you? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?
On 7/16/06, Martin Joseph [EMAIL PROTECTED] wrote: On Jul 16, 2006, at 11:36 AM, Maxim Vexler wrote: Hello list I'm trying to setup asterisk as an answering machine. How can I set asterisk to Answer() incoming call ONLY after specified count of ring cycles ? In the current situation I have the PBX connected to a home line, where POTS device are also connected on the same circuit. What I'm trying to do is allow a grace period where a POTS device could be picked up and those stop the ring indication on the line by this causing asterisk to not answer the call. In present situation even if the incoming phone call is taken off hook by a POST device asterisk still starts playing its incoming call IVR after the specified(where?) number of seconds. I don't think you can do that, since asterisk has no way to know when the shared PSTN line is answered by your analog phones... I don't think asterisk counts the rings, as much as it waits for answered status, which it is never going to see in your current configuration. I am a relative newb though, so maybe someone else here has a brilliant idea for you? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You have a point but no way am I going to accept that as an answer. Here's the log off such case : Jul 16 21:59:20 DEBUG[4387] chan_zap.c: Monitor doohicky got event Ring Begin on channel 1 Jul 16 21:59:21 DEBUG[4387] chan_zap.c: Monitor doohicky got event Ring/Answered on channel 1 Jul 16 21:59:21 DEBUG[4387] dsp.c: dsp busy pattern set to 0,0 Jul 16 21:59:21 VERBOSE[4411] logger.c: -- Starting simple switch on 'Zap/1-1' Jul 16 21:59:21 DEBUG[4373] devicestate.c: Changing state for Zap/1 - state 2 (In use) Jul 16 21:59:21 DEBUG[4412] app_queue.c: Device 'Zap/1' changed to state '2' (In use) Jul 16 21:59:29 WARNING[4411] chan_zap.c: CallerID returned with error on channel 'Zap/1-1' Jul 16 21:59:29 DEBUG[4411] pbx.c: Launching 'Answer' Jul 16 21:59:29 VERBOSE[4411] logger.c: -- Executing Answer(Zap/1-1, ) in new stack Jul 16 21:59:29 DEBUG[4411] chan_zap.c: Took Zap/1-1 off hook Jul 16 21:59:29 DEBUG[4411] chan_zap.c: Enabled echo cancellation on channel 1 Jul 16 21:59:29 DEBUG[4411] chan_zap.c: No echo training requested Jul 16 21:59:29 DEBUG[4411] pbx.c: Launching 'Set' As you can see, the first two events are event Ring and event Ring/Answered. What I need is the driver of chan_zap.c counting 5 event Ring before starting Ring/Answered. It can't be that hard (I think). Thank you for your answer. -- Cheers, Maxim Vexler Free as in Freedom - Do u GNU ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi 'Unable to Find Key'
Hi, I got this also and actually I still get this message. But i did realise the setup you ar trying to realise now. I wrote a little document on how to achieve this with trixbox. You can find it here:
http://kneh.xs4all.nl/tijmen/asterisk/Using%20DUNDi%20with%20Trixbox.pdfNote that you should replace password with ${SECRET}, in the mappings part of dundi.conf, to increase securityHope this helps you.
On 7/14/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I'm trying to trunk calls from one asterisk box to another with DUNDi.The following is appearing in my /var/log/asterisk/messages file:Jul 13 17:29:59 DEBUG[25674] db.c: Unable to find key '000E0CA1926F/oe_tech/180app/e' in family 'dundi/cache'
Jul 13 17:29:59 DEBUG[25674] db.c: Unable to find key '000E0CA1926F/oe_tech/180app/e' in family 'dundi/cache'Jul 13 17:29:59 DEBUG[25674] db.c: Unable to find key '000E0CA1926F/oe_tech/180app/r' in family 'dundi/cache'
Jul 13 17:29:59 DEBUG[25674] db.c: Unable to find key 'hint/000E0CA1926F/o/180app/e' in family 'dundi/cache'Jul 13 17:29:59 DEBUG[25674] db.c: Unable to find key 'hint/000E0CA1926F/o/180app/e' in family 'dundi/cache'
Jul 13 17:29:59 DEBUG[25674] db.c: Unable to find key 'hint/000E0CA1926F/o/180app/r' in family 'dundi/cache'Jul 13 17:29:59 DEBUG[25674] db.c: Unable to find key 'hint/000E0CA1926F/oe/180app/e' in family 'dundi/cache'
Jul 13 17:29:59 DEBUG[25674] db.c: Unable to find key 'hint/000E0CA1926F/oe/180app/e' in family 'dundi/cache'Jul 13 17:29:59 DEBUG[25674] db.c: Unable to find key 'hint/000E0CA1926F/oe/180app/r' in family 'dundi/cache'
and so on...What the heck is going on? At one point, I removed my astdb file on one of the systems, which may have goofed things up. I've since removed astdb on all three asterisk boxes, and done a 'dundi flush' also, but the messages keep re-appearing.
When dialling the number with IAX, and this problem occurs, Asterisk completely goes out to lunch. Most console commands no longer respond.Anyone got any ideas?Doug___
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Re: [asterisk-users] Re: Wrong account code from iax_buddies
On 16 Jul 2006, at 19:00, voiplist wrote: Trixbox is only used as the client to simulate what we already saw happening with a customer. I don't think the fact that we used a Trixbox on the client side has anything to do with the problem on the server side which is not using Trixbox. The on the server side Asterisk only sees the Trixbox as a client coming in under an iax_buddies record which has an accountcode assigned to it. The server side is taking the call, then inserting the CDR with a correct account code, dst, src and so on but the channel is wrong. I still think that if you want your problem fixed, you'll need to reduce it to a test case that an asterisk maintainer can reproduce, without using trixbox. (or fix it yourself) The other alternative would be to capture some IAX traces either with ethereal or asterisk's iax2 debug of a call where this goes wrong. This might give clues as to how your problem happens. If you do produce IAX traces I'd be happy to look them over as I'm going to likely run into this problem myself in a couple of weeks. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Automation of call initiation
Hi, I need to monitoran asterisk server, so planning to use some tools which can initiate call to a number (for asterisk server)periodically and can interpret the response, is anything as such already available?? or any pointer?? thanks in advance Nitin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue RoundRobin
Hi, I'm setting up a new asterisk for an ecommerce company with cust sup dept. The problem I'm having is with Roundrobin (and rrmemory also): Let's suppose that I have 2 agents logged in into a queue. When a client calls, and both agents are available. It rings the first one, but it doesn't answer the phone. The timeout takes effect and it should start ringing the second agent. But it doesn't. It keeps ringing the first one until it answers the phone Here's my queue.conf: [general] [QueueEN] announce = ann-english strategy = rrmemory timeout = 5 retry = 1 wrapuptime=0 maxlen = 0 announce-frequency = 20 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 [QueueES] strategy = rrmemory timeout = 5 retry = 5 wrapuptime=0 maxlen = 0 announce = ann-spanish announce-frequency = 10 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 The timeout is set too low so the test is faster. Cheers, Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Wrong account code from iax_buddies
Actually, at this point this info was more for the community as a whole. We don't need to fix this now because the account code is right and that's what is important. On 7/16/06, Tim Panton [EMAIL PROTECTED] wrote: On 16 Jul 2006, at 19:00, voiplist wrote: Trixbox is only used as the client to simulate what we already saw happening with a customer. I don't think the fact that we used a Trixbox on the client side has anything to do with the problem on the server side which is not using Trixbox. The on the server side Asterisk only sees the Trixbox as a client coming in under an iax_buddies record which has an accountcode assigned to it. The server side is taking the call, then inserting the CDR with a correct account code, dst, src and so on but the channel is wrong. I still think that if you want your problem fixed, you'll need to reduce it to a test case that an asterisk maintainer can reproduce, without using trixbox. (or fix it yourself) The other alternative would be to capture some IAX traces either with ethereal or asterisk's iax2 debug of a call where this goes wrong. This might give clues as to how your problem happens. If you do produce IAX traces I'd be happy to look them over as I'm going to likely run into this problem myself in a couple of weeks. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom config file location
Stephen Murphy wrote: My question is: How do I get the current config files the phone is using off the phone? AFAIK, you can't. :( You can only provide new configuration files from your FTP/TFTP server. However, the Polycoms do strange things when they've been configured in multiple locations. You might find the phone overwriting the configuration files with its original configuration. That is not confirmed though. I've just seen my Polycoms do weird stuff in the wild. :) -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: 1 300 SQUIZ (77859) Fitzroy, VIC T: 03 9235 5400 3065 F: 03 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Regression testing dialplan changes
Hi all, As I am starting to have stable live releases of a dialplan and development work going on in parallel I need to have some sort of regression test in place to ensure that no key functions of the current dialplan are broken by a new version. Does anyone have pointers to the best way to run an automated test on the dialplan, what I am really hoping for is something that looks and works a bit like an nUnit type automated test but I'll take almost any automated testing approach over the alternative - which is manual re-testing every time we change a dialplan. Thanks, Nic ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re: [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!
So I can just install it over 1.2.9? This is what I did and everything seems to be working fine. Date: Sat, 15 Jul 2006 21:47:40 +1200 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released! -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Julian Varanini wrote: What is the best way to update from 1.2.9 to 1.2.10? If it was downloaded from SVN then you can just type make update in the directory. If it was a .tar.gz download then you will need to reinstall. I would recommend using the 1.2 branch of SVN as it means you don't have to wait for the releases to get the bugfixes. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEuLm8S6d5vy0jeVcRAk9RAJ478UyMx8g7WLzkhAp+9VT9eZfXewCggHXo 9bn2Ob7u9jlDsqrKLZVrv/4= =y79J -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom IP301 and Queues
Is there any way to use the polycom phones to log into and out of queues? So the polycom phone could show their current status in that queue? logged in / logged out for example. Thanks Julian Subject: RE: [asterisk-users] PRI dropouts From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Sat, 15 Jul 2006 20:47:17 +1000 Hmm - I have had 2 bad PRI installs out of about 20, and both times it was faulty wiring from the Telco. But getting them to fix it can be a real struggle! Paul Hales Technical Manager www.asteriskit.com.au On Sat, 2006-07-15 at 12:23 +1000, James Sturges wrote: Have had L O T S of trouble like this, the settings zap config files seem to have to e exact, please send email to [EMAIL PROTECTED] and I will send config files. Thanks James __ From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Withnall Sent: Saturday, 15 July 2006 11:05 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] PRI dropouts Recently we cut over to using asterisk (trixbox 1.1.1) for our production system. We are using a TE110P digium card (Primary rate) with a Telstra onramp 10. Sometimes when people call, on their end it doesn’t seem to connect. On our end, we get caller id, it passes ok to the sip phone but then no-one is there. Anyone have any similar problems and worked out how to solve it ? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue RoundRobin
Hi Santiago, Unless it is a typo on the wiki, I think you want your queue.conf to be like this: member = Agent/@1 member = Agent/:2,1 That way you include group 1, and then include group 2 with consideration of penalty. From the problem you are having it sounds like the agent whose phone keeps ringing is in a lower penalty then the other agent. Are both agents in the same group? If you make the one agent busy does it ring to the next phone? If not, what does the CLI say when it tries to connect the next call to the second phone? Kevin Santiago del Castillo wrote: Hi, I'm setting up a new asterisk for an ecommerce company with cust sup dept. The problem I'm having is with Roundrobin (and rrmemory also): Let's suppose that I have 2 agents logged in into a queue. When a client calls, and both agents are available. It rings the first one, but it doesn't answer the phone. The timeout takes effect and it should start ringing the second agent. But it doesn't. It keeps ringing the first one until it answers the phone Here's my queue.conf: [general] [QueueEN] announce = ann-english strategy = rrmemory timeout = 5 retry = 1 wrapuptime=0 maxlen = 0 announce-frequency = 20 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 [QueueES] strategy = rrmemory timeout = 5 retry = 5 wrapuptime=0 maxlen = 0 announce = ann-spanish announce-frequency = 10 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 The timeout is set too low so the test is faster. Cheers, Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP301 and Queues
Hi Julian, If the 301's support ACD log in and log out, they should display a soft button showing the current status of the phone, I know for sure the 601's do. Personally with our 601's I used two of the contact lines and made my own log in and logout buttons and wrote my own script to log our agents in. It doesn't display the status, but I have a section on our intranet page showing the status of all members of a queue that are logged in. So it may not be the answer you wanted, and again I don't have any experience with the 301's to say what they can and cannot do, but there are some workarounds that will come close to the same goal. kevin Julian Varanini wrote: Is there any way to use the polycom phones to log into and out of queues? So the polycom phone could show their current status in that queue? logged in / logged out for example. Thanks Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Regression testing dialplan changes
Y'know, I was thinking about a similar idea recently, primarily because I do a lot of work with dialplan based apps. It would be great if there was a way to set up a _complete_ call (meaning it would include what digits to enter when, etc) in a test and then run it against the dialplan being worked on. One thing that I have found is really helpful is coding your dialplans in AEL2 and running the aelparse program with the -d option, which uses the CWD as the base to read from. The parser tells you when something's not right and tells you where it found the error. However this isn't helpful for things that only take place during a call like database lookups and variable values. I'm interested to see the answers that come from your post, thanks for putting it out there! Sherwood McGowan Consultant -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nic Hughes Sent: Sunday, July 16, 2006 6:31 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Regression testing dialplan changes Hi all, As I am starting to have stable live releases of a dialplan and development work going on in parallel I need to have some sort of regression test in place to ensure that no key functions of the current dialplan are broken by a new version. Does anyone have pointers to the best way to run an automated test on the dialplan, what I am really hoping for is something that looks and works a bit like an nUnit type automated test but I'll take almost any automated testing approach over the alternative - which is manual re-testing every time we change a dialplan. Thanks, Nic ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP configuration by group
Sharon Lim wrote:
Hi there,
I would like to ask, is it possible to group sip user? Means group A with
sip user 100,200 and group B with sip user 100,200?
thanks in advance.
in your dialplan, define the following variables:
GROUP_A=SIP/100SIP/200
GROUP_B=SIP/150SIP/200
and in your dial string
exten = blah,1,Dial(${GROUP_A})
exten = moreblah,1,Dial(${GROUP_B})
Flynn
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[asterisk-users] Vicidial + Unicall mfcr2
Does Vicidial work together with Unicall/mfcr2 ? Best Regards-- Bruno de Assumpção Loureiromsn: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PRI dropouts - solution I hope...
In my experience PRI pass through setups have been false economy. They seem to save a few dollars, but you still have to spend the money to save it and they never run as well. Paul Hales -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au bus: 03 8320 8100 mob: 0434 673 529 On Sat, 2006-07-15 at 22:22 +1000, James Sturges wrote: Hi, had a few ask for this so thought may be of interest to the list. This is actually for the following setup: Telstra ISDN30 - Asterisk - BP250 PABX The ISDN10, 20, 30’s are all the same physical link, but you may need to change the bchan and dchan settings for ISDN 10 or 20. We have had lot of issues over 12 months, including physical cable issues, etc. But this config has passed Telstra test equipment both on site and in the exchange. The calls dropping out (for us) are timing issues do to telling Asterisk to gets it synch from the Telstra line and providing synch to the PABX. Anyway, Here it is, does not look like much but have had experts working on it for a while. The system handles 1800 – 2000 calls per day. Thanks James ZAPATA.conf [channels] context=default musiconhold=default switchtype=euroisdn usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=0.0 txgain=0.0 group=1 context=te405p-intelstra pridialplan=local signalling=pri_cpe callerid=asreceived channel=1-15, 17-31 group=4 context=te405p-frombp250 pridialplan=local signalling=pri_net overlapdial=yes callerid=asreceived channel=94-108, 110-124 ZAPTEL.conf # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,0,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 span=3,0,0,ccs,hdb3,crc4 bchan=63-77 dchan=78 bchan=79-93 span=4,0,0,ccs,hdb3,crc4 bchan=94-108 dchan=109 bchan=110-124 loadzone=au defaultzone=au ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP configuration by group
hmm...the group functions is to dial all the sip account, right. assuming if the dial plan is like exten = blah,1,Dial(${GROUP_A})exten = moreblah,1,Dial(${GROUP_B})then it will dial sip100 sip200 at the same time right? But i want to group it as different company. Is it possible?
Assuming, if 1 have 2 company and want to have same sip account context, how do i differentiate with it? Thanks in advance. On 7/17/06, El Flynn
[EMAIL PROTECTED] wrote:
Sharon Lim wrote: Hi there, I would like to ask, is it possible to group sip user? Means group A with sip user 100,200 and group B with sip user 100,200? thanks in advance.
in your dialplan, define the following variables:GROUP_A=SIP/100SIP/200GROUP_B=SIP/150SIP/200and in your dial stringexten = blah,1,Dial(${GROUP_A})exten = moreblah,1,Dial(${GROUP_B})
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Re: [asterisk-users] Vicidial + Unicall mfcr2
I don't know as I've never tested libunicall on any Asterisk system. VICIDIAL will currently only work with Zap/SIP/IAX channels. Can you install Unicall to use with USA T1s? Would it make sense to do so for any practical purposes? What does the Unicall channel show up as inside of asterisk? (Zaptel channels show up as Zap/1-1 and so on.) MATT--- On 7/16/06, Bruno de Assumpção Loureiro [EMAIL PROTECTED] wrote: Does Vicidial work together with Unicall/mfcr2 ? Best Regards -- Bruno de Assumpção Loureiro msn: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE
I have a customer witha Polycom 501 phone behind a NAT. His phone is connected tohis Netgear router at home which in turn is connected to his cable modem. The phone is set up to register with our remote Asterisk server which is on a public, static IP address, with no NAT. If we set qualify=yes, our Asterisk console shows his extension becoming UNREACHABLE for a minute, then REACHABLE for a minute, then UNREACHABLE again, in an endless cycle. If we try to call the phone while it is UNREACHABLE, the phone never rings and the call goes straight to voice mail.This is very annoying. If we set qualify=no, then if we try to call the phone, the phone sometimes does not ring at all, and we hear silence. The call eventually goes to voice mail. This is equally annoying to the customer. What is the solution to this problem? We have other customers with Polycom phones behind NAT, and they don't have this problem. Will we have better luck if we replace the Polycom with a Linksys 942 phone? Here is some console output: Jul 16 21:44:24 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE! Last qualify: 174Jul 16 21:45:33 NOTICE[19981]: chan_sip.c:9697 handle_response_peerpoke: Peer '280' is now REACHABLE! (3181ms / 5000ms) Jul 16 21:47:37 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE! Last qualify: 175 Here is the way the phone is set up in sip.conf: [280]type=peerusername=280secret=280host=dynamicdtmfmode=rfc2833callerid=John 280context=company_xmailbox=280nat=yescanreinvite=noqualify=5000We are using Asterisk 1.2.5 with standard .conf files. We are not using realtime or databases. Any help would be highly appreciated. Rana Dutt Softel Solutions [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vicidial + Unicall mfcr2
On 7/17/06, Matt Florell [EMAIL PROTECTED] wrote:
I don't know as I've never tested libunicall on any Asterisk system.VICIDIAL will currently only work with Zap/SIP/IAX channels.Can you install Unicall to use with USA T1s?
No , it works with MFC/R2 signaling . There are some countries where ISDN it is not the only choice.
Would it make sense to doso for any practical purposes?
If I have to change mfcr2 to ISDN the PSTN will change my line number.
What does the Unicall channel show up as inside of asterisk? (Zaptelchannels show up as Zap/1-1 and so on.)
Unicall/1-1
extensions.conf
exten = 38521717,1,Dial(Unicall/g1/${EXTEN},,tT)
MATT---
I think it works...
On 7/16/06, Bruno de Assumpção Loureiro
[EMAIL PROTECTED] wrote: Does Vicidial work together with Unicall/mfcr2 ?Best Regards -- Bruno de Assumpção Loureiro msn:
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Bruno de Assumpção Loureiromsn: [EMAIL PROTECTED]
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Re: [asterisk-users] Injecting prerecorded audio into active call
Yeah a bit messy I guess. I had been hoping for a simple solution, but knew there most likely wasn't! The one idea I did have would be to use some kind of SIP api on the web backend. Then bring the backend extension into a conference, then from the web api you would have to select the call to play audio in. This idea would work well I think, as it would mean the system can be use regardless of the training call being active on the asterisk box, as long as their system supported conference calls. This is where I fall down though, I'm no developer! Anyone know of an api that would allow this? Cheers Nick John covici wrote: Well, if the web interface copied the call to a standard name and you had an extension using Playback or ControlPlayback to play that file and then bridged the call -- maybe that wold work -- much of a kludge though. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE
According to your console output it looks like there is some major latency. What is the average ping time from your asterisk machine to the polycom phone? - Original Message - From: Rana Dutt To: Asterisk Users Sent: Sunday, July 16, 2006 6:51 PM Subject: [asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE I have a customer witha Polycom 501 phone behind a NAT. His phone is connected tohis Netgear router at home which in turn is connected to his cable modem. The phone is set up to register with our remote Asterisk server which is on a public, static IP address, with no NAT. If we set qualify=yes, our Asterisk console shows his extension becoming UNREACHABLE for a minute, then REACHABLE for a minute, then UNREACHABLE again, in an endless cycle. If we try to call the phone while it is UNREACHABLE, the phone never rings and the call goes straight to voice mail.This is very annoying. If we set qualify=no, then if we try to call the phone, the phone sometimes does not ring at all, and we hear silence. The call eventually goes to voice mail. This is equally annoying to the customer. What is the solution to this problem? We have other customers with Polycom phones behind NAT, and they don't have this problem. Will we have better luck if we replace the Polycom with a Linksys 942 phone? Here is some console output: Jul 16 21:44:24 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE! Last qualify: 174Jul 16 21:45:33 NOTICE[19981]: chan_sip.c:9697 handle_response_peerpoke: Peer '280' is now REACHABLE! (3181ms / 5000ms) Jul 16 21:47:37 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE! Last qualify: 175 Here is the way the phone is set up in sip.conf: [280]type=peerusername=280secret=280host=dynamicdtmfmode=rfc2833callerid="John" 280context=company_xmailbox=280nat=yescanreinvite=noqualify=5000We are using Asterisk 1.2.5 with standard .conf files. We are not using realtime or databases. Any help would be highly appreciated. Rana Dutt Softel Solutions [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.394 / Virus Database: 268.10.1/389 - Release Date: 7/14/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP configuration by group
At 06:25 PM 7/16/2006, you wrote:
exten = blah,1,Dial(${GROUP_A})
exten = moreblah,1,Dial(${GROUP_B})
then it will dial sip100 sip200 at the same time right? But i want
to group it as different company. Is it possible?
Assuming, if 1 have 2 company and want to have same sip account
context, how do i differentiate with it?
One of these for each business should help if you have callerID
display on your phones.
exten = s,n,GoToIf($[${CALLERID(Name)} = ]?noCID:prefixCID)
exten = s,n(prefixCID),Set(CALLERID(Name)=B1_${CALLERID(Name)})
exten = s,n,goto(doneCID)
exten = s,n(noCID),Set(CALLERID(Name)=Business1)
exten = s,n,goto(answercall)
Ira
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[asterisk-users] call forwarding
Hi people. I want to know about call forwarding. I dial *72, and a message say me to dial the extension , I did, then the message said is forward is UNCONDITIONLA . But when I call , it doesn't work the forwarding. Who can help me please. Best Regards Ever ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Skype protocol cracked?
Hi CF I find that yes. The model of skipe was cracked.See link below: http://politics.slashdot.org/politics/06/07/14/1514226.shtml 2006/7/16, C F [EMAIL PROTECTED]: http://news.yahoo.com/s/zd/20060714/tc_zd/183411 ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SRTP enabling
Hello,In some countries i found that they are blocking SIP port 5060so instead of this i change to another port 1221, and its workwell. But in one country the are not blocking SIP but they areplaying with RTP packets, if they filtered it is VoIP RTP theyare doing something called party cannot hear or some time callercannot hear but called party can hear well.So i cosider to use SRTP to make encryption. and i am usingmy asterisk in VPS so i have full control to manage the server.If you guys have better Idea to prevent such kind of issue, itwill be good for us.AbdulMost of the blocking in other countries, was not for RTP traffic, but for signaling traffic (SIP usually, Mexico x Vonage comes to mind). You are sure they are blocking RTP traffic ? And, from what I understand, in some places the gov. forced the ISPs to remove the blocking (at least, I heard of one such a case in Brazil, a DSL provider started to block SIP, and Anatel, Brazil gov. entity that regulate telephony and others, asked them to remove the blocking, others with more knowledge of the case may be able to add their remarks) Blocking SIP if you control the server is somewhat easy to prevent (if is a plain dumb UDP port 5060 filtering), just have your server listen in another UDP port... See the all-new, redesigned Yahoo.com. Check it out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SRTP enabling
On Jul 16, 2006, at 9:45 PM, Abdul wrote: Hello, In some countries i found that they are blocking SIP port 5060 so instead of this i change to another port 1221, and its work well. But in one country the are not blocking SIP but they are playing with RTP packets, if they filtered it is VoIP RTP they are doing something called party cannot hear or some time caller cannot hear but called party can hear well. So i cosider to use SRTP to make encryption. and i am using my asterisk in VPS so i have full control to manage the server. If you guys have better Idea to prevent such kind of issue, it will be good for us. Why not use IAX2? Then you only have one port to worry about reconfiguring ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sphinx and Asterisk Integration, How?
After several hours of searching the Internet, couldn't understand how can I integrate Asterisk with Sphinx voice recognition system. The sphinx software itself I've installed on my server. I need help from those who have successfully done it and can guide me how to do it. Thanks-- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel on dual processor, How?
I am trying to install zaptel on dual Xeon processor but it gives error, saying 'You do not appear to have the kernel sources for your current kernel installed.make: *** [linux26] Error 1' Googled for many hours, but nothing, except to use non smp kernel. How can I build zaptel for smp.-- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel on dual processor, How?
On Monday 17 July 2006 12:05 am, Zeeshan Zakaria wrote: I am trying to install zaptel on dual Xeon processor but it gives error, saying 'You do not appear to have the kernel sources for your current kernel installed. make: *** [linux26] Error 1' Googled for many hours, but nothing, except to use non smp kernel. How can I build zaptel for smp. Install your kernel sources the process will vary depending on your distro -- Dennis Gilmore, RHCE Proud Australian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel on dual processor, How?
You will have to install the kernel sources - what distro are you running? PaulH On Mon, 2006-07-17 at 01:05 -0400, Zeeshan Zakaria wrote: I am trying to install zaptel on dual Xeon processor but it gives error, saying 'You do not appear to have the kernel sources for your current kernel installed. make: *** [linux26] Error 1' Googled for many hours, but nothing, except to use non smp kernel. How can I build zaptel for smp. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel on dual processor, How?
How to install kernel sources? On 7/17/06, Dennis Gilmore [EMAIL PROTECTED] wrote: On Monday 17 July 2006 12:05 am, Zeeshan Zakaria wrote: I am trying to install zaptel on dual Xeon processor but it gives error, saying 'You do not appear to have the kernel sources for your current kernel installed. make: *** [linux26] Error 1' Googled for many hours, but nothing, except to use non smp kernel. How can I build zaptel for smp.Install your kernel sourcesthe process will vary depending on your distro--Dennis Gilmore, RHCEProud Australian___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel on dual processor, How?
On Mon, Jul 17, 2006 at 12:06:20AM -0500, Dennis Gilmore wrote: On Monday 17 July 2006 12:05 am, Zeeshan Zakaria wrote: I am trying to install zaptel on dual Xeon processor but it gives error, saying 'You do not appear to have the kernel sources for your current kernel installed. make: *** [linux26] Error 1' Googled for many hours, but nothing, except to use non smp kernel. How can I build zaptel for smp. Install your kernel sources the process will vary depending on your distro (or rather: kernel headers for your current kernel configuration) What is the ourput of 'uname -r' ? What is your distriution? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Regression testing dialplan changes
On Sun, Jul 16, 2006 at 11:31:24PM +0100, Nic Hughes wrote: Hi all, As I am starting to have stable live releases of a dialplan and development work going on in parallel I need to have some sort of regression test in place to ensure that no key functions of the current dialplan are broken by a new version. Does anyone have pointers to the best way to run an automated test on the dialplan, what I am really hoping for is something that looks and works a bit like an nUnit type automated test but I'll take almost any automated testing approach over the alternative - which is manual re-testing every time we change a dialplan. Do you expect those tests to run on a running Asteris system or a copy of the configuration file? What do you want to test, exactly? 'show dialplan' can show the parsed configuration (also after AEL include files expansion). This can help validateproper configuration generation. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
