[Asterisk-Users] Asterisk and Hipath interconnections
Hello Can anyone know where may download chan_cornet for interconnection Asterisk and Hipath via IP Thanks Viktor ___ --Bandwidth 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] Ringing Delay
Hi Chan, 1/ be sure to have correctly inputed your country zone 2/ disable the fax recognition in zapata.conf Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de chan (Alpha Trilogies Networls) Envoyé : lundi 27 février 2006 08:35 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Ringing Delay Hi, Can some one advice me that how can I make the FXO channels port answer an incoming calls, means when I call from Lan line to Asterisk TDM400, my phone get ring immediately. When POT FXO port is ringing, Asterisk seems like studying the incoming ringing pattern even it did answer the call. I did not activate the usedestingtive, but why it seems delaying an incoming calls? Normal PBX, say will only delay 1 cycle as max in analog line, but Asterisk is about 2 sec...??? ___ --Bandwidth 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] Newbie config help? Wellgate 3701a (answers)
Short version: Flash device with latest SIP firmware (currently 1.04) Set Network (I am using the LAN port only) and SIP config as expected. Set Line configuration so that the FXO is hotline to the asterisk extension you want to ring with incoming PSTN calls (mine is set to 2020). Set System configuration so that the keypad type is inband (rfc2833 doesn't seem to work?). Change the Routing Table so that the default for IP is set to FXO for destination. Click commit data and then the commit button. Click reboot and then the reboot button. Asterisk looks like this: ; ; SIP entry for user Wellgate (FXO) [2003] type=friend secret=hushhush dtmfmode=inband auth=md5 host=dynamic nat=yes reinvite=no canreinvite=no disallow=all allow=ulaw context=autocontext callerid=Alton Qwest Line2065551212 ; ; SIP entry for user Wellgate (FXS) [2005] type=friend secret=Sh auth=md5 host=dynamic disallow=all allow=ulaw allow=g729 allow=alaw allow=gsm allow=ilbc context=autocontext callerid=Alton Estates2005 And the dialplan bit: ; Dial any 7 digit numbers through that plain old telephone network exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _NXX,2,Hangup ; Still a few minor issue 1) with double ringback on IAXCOMM, and one with the beginning of audio being snipped on the FXS connected phone? Not too bad though for a newb with a couple of 1/4 days :~) I think it fixes my echo issue also. I can hear a sort of crackle for the first 3 seconds of the call and then it's all good. ___ --Bandwidth 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] Skype vs. an Xlite registered to Asterisk
2006/2/26, hugolivude [EMAIL PROTECTED]: I have a bunch of road warriors who I've set up with Xlite clients. Unfortunately the sound quality has been intermittent at best. What codec dis you use?? I think xlite support speex, that is the better codec I've tested when connections are under hevy traffic (p2p applications). G729 is good too, but Speex really worked great in my tests. Sometimes I was thinking about trying an Xlite client that can support G729. Anyone had experience with that? Does it significantly improve voice quality? What you need to improve (or decress) is the bandwidth usage. Check this: http://www.voip-info.org/wiki/view/Bandwidth+consumption but try speex if it is supported by your sip phone. It is free, variable bitrate and adapts to the available bandwidth, (it is based on ogg codec). -- Alejandro Vargas ___ --Bandwidth 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] Prepaid / postpaid solution
Alexander Burke wrote: At 05:03 PM 02/26/2006, you wrote: I want to match the user from the users callerid. All users have DIDs. You probably shouldn't do that for security reasons -- rather, match them according to the SIP username/password pair they provide when they register. Hm, Maybe you're right. The Idea is to get the same solution, The user is automaticlly identified in the billingsystem... /Mike ___ --Bandwidth 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 and Hipath interconnections
Hi Victor Looking for the same answers here too. We are regional distributors for Hicom HiPath in this part of the world and until now we are still waiting for chan_cornet to come around. So far we have successfully interconnected via BRI (mISDN) and PRI (Zaptel) and it works great. Let's see if it's too good to be true soon. Best regards, Stephen Viktor Tatianin wrote: Hello Can anyone know where may download chan_cornet for interconnection Asterisk and Hipath via IP Thanks Viktor ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Hipath interconnections
hi all, maybe i am mistaken but it seems to me that the HiPath 2000 series is an Asterisk based system. why am i saying this? because Siemens announce it is a Linux, Open Source system. so, as i do not know any OTHER PBX Linux- Open Source system rather then Asterisk, does anybody know something else? otherwise, if it is an Asterisk system, so why there is a need for Cornet? you can interconnect with IAX, isn't it? Mickey On 2/27/06, Stephen Arulraj [EMAIL PROTECTED] wrote: Hi VictorLooking for the same answers here too. We are regional distributors forHicom HiPath in this part of the world and until now we are still waiting for chan_cornet to come around. So far we have successfullyinterconnected via BRI (mISDN) and PRI (Zaptel) and it works great.Let's see if it's too good to be true soon.Best regards,Stephen Viktor Tatianin wrote:HelloCan anyone know where may download chan_cornet for interconnection Asteriskand Hipath via IPThanksViktor___ --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 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] Asterisk and Hipath interconnections
Hi, if yo are looking a way to interconnect Asterisk with a HiPath 4000 via IP, so you have 2 ways to do it. - via oh323 (for HiPath 4000 version 1 and 2) - since HiPath4000 version 3 you are able to interconnect using sipQ (SIP Trunking) --- Viktor Tatianin [EMAIL PROTECTED] wrote: Hello Can anyone know where may download chan_cornet for interconnection Asterisk and Hipath via IP Thanks Viktor ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ 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 and Hipath interconnections
Hi again, i don't think that the HiPath2000 is an Asterisk based system. AFAIK the HiPath2K is only configurable using a Web-based tool (no console access). For the moment the HiPath2K will only be release with CornetIP (HFA). No SIP (panned in a second step) and unfortunazely no IAX are avalaible. so if teh HiPath2K is an Asterisk based PBX, it meens that Siemens has developped a pseudo chan_cornet... but i don't think so... --- Tele Cost Price Reducer [EMAIL PROTECTED] wrote: hi all, maybe i am mistaken but it seems to me that the HiPath 2000 series is an Asterisk based system. why am i saying this? because Siemens announce it is a Linux, Open Source system. so, as i do not know any OTHER PBX Linux- Open Source system rather then Asterisk, does anybody know something else? otherwise, if it is an Asterisk system, so why there is a need for Cornet? you can interconnect with IAX, isn't it? Mickey On 2/27/06, Stephen Arulraj [EMAIL PROTECTED] wrote: Hi Victor Looking for the same answers here too. We are regional distributors for Hicom HiPath in this part of the world and until now we are still waiting for chan_cornet to come around. So far we have successfully interconnected via BRI (mISDN) and PRI (Zaptel) and it works great. Let's see if it's too good to be true soon. Best regards, Stephen Viktor Tatianin wrote: Hello Can anyone know where may download chan_cornet for interconnection Asterisk and Hipath via IP Thanks Viktor ___ --Bandwidth 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 __ 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] Ringing Delay
Hi, I did change the RING parameters to my country, but seems like no improvement, so how to confirm the ringing frequency than from Telco, any device to test it out? Date: Mon, 27 Feb 2006 09:28:15 +0100 From: [EMAIL PROTECTED] Subject: RE : [Asterisk-Users] Ringing Delay Hi Chan, 1/ be sure to have correctly inputed your country zone 2/ disable the fax recognition in zapata.conf Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de chan (Alpha Trilogies Networls) Envoyi : lundi 27 fivrier 2006 08:35 @ : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Ringing Delay Hi, Can some one advice me that how can I make the FXO channels port answer an incoming calls, means when I call from Lan line to Asterisk TDM400, my phone get ring immediately. When POT FXO port is ringing, Asterisk seems like studying the incoming ringing pattern even it did answer the call. I did not activate the usedestingtive, but why it seems delaying an incoming calls? Normal PBX, say will only delay 1 cycle as max in analog line, but Asterisk is about 2 sec...??? ___ --Bandwidth 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] Zap tuning for echo/gain
I'm having a bit of an issue with one of the bargain x100p clones, and I'm not sure what the right approach is. My symptom started as way loud offset delayed echo from voip hardphones - PSTN through the clone card. I played with and then learned everything I could about echo cancelling, and have managed to bring the echo down to acceptible levels, where it's not too intrusive. In my travels I discovered ztmonitor, and thought I'd run this. Now, with no call and no activity on the line, the RX side shows a reading of about 1/3 of the bar graph. I figured this is bad (right?) and tweaked my rxgain value (I had never had to touch this before) until ztmonitor only showed one bar. I accomplished this by adding a negative value to the rxgain, I believe it was -12. My question is: Is this the right approach? Should I be tweaking my gains first, and then adjusting echo, or vice versa? Is there something else completely I should be looking at first, or is this the right thing to do? I did have a genuine Digium analog TDM card in there previously, but I had to swap boxes and for some reason the TDM isn't probing (and digium support haven't been too helpful, but that is another story). The TDM didn't need any tweaks; it seemed to have worked out of the box (although I might have been more tolerant of slight echo before, but the clone card was simply intolerable for conversation). Any suggestions or comments welcome. Thanks, J. ___ --Bandwidth 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 and Hipath interconnections
Hi Mickey Seeing is beliving. Any clues to your claims? Sstephen Tele Cost Price Reducer wrote: hi all, maybe i am mistaken but it seems to me that the HiPath 2000 series is an Asterisk based system. why am i saying this? because Siemens announce it is a Linux, Open Source system. so, as i do not know any OTHER PBX Linux- Open Source system rather then Asterisk, does anybody know something else? otherwise, if it is an Asterisk system, so why there is a need for Cornet? you can interconnect with IAX, isn't it? Mickey On 2/27/06, Stephen Arulraj [EMAIL PROTECTED] wrote: Hi Victor Looking for the same answers here too. We are regional distributors for Hicom HiPath in this part of the world and until now we are still waiting for chan_cornet to come around. So far we have successfully interconnected via BRI (mISDN) and PRI (Zaptel) and it works great. Let's see if it's too good to be true soon. Best regards, Stephen Viktor Tatianin wrote: Hello Can anyone know where may download chan_cornet for interconnection Asterisk and Hipath via IP Thanks Viktor ___ --Bandwidth 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 ___ --Bandwidth 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] IAX provider recommendation wanted
Hi, Can someone recommend an IAX provider for US DIDs who will: 1) Accept Canadian credit cards (rules out Junction Networks!) 2) Can do local number porting (LNP) 3) Have great audio quality I tried Teliax, but the IAX audio quality was terrible - pops and clicks galore! The Teliax SIP quality was better, but still horrible compared to my Canadian DID IAX provider, Unlimitel.ca. - Mike ___ --Bandwidth 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] Problems dialing to another Asterisk server
Hi,
I have a problem dialing a SIP phone which is logged in as different
Astesrik machine from the one I am working with.
I want to call a phone in Another astersik machine in , if it answers,
calling a SiP phone registered in my ASterisk:
My dialplan is:
[mariaSIP]
exten = _1.,1,Wait(1)
exten = _1.,2,Dial(SIP/[EMAIL PROTECTED]:5038,20)
exten = _1.,3,HangUp()
exten = 222,1,MusicOnHold()
exten = 444,1,Dial(${STRING3})
exten = 444,2,Ringing
SIP/6021 is the telephone logged in the another machine, which is
192.168.0.51 and asterisk is listening in 192.168.0.51, port 5038
And my Manager-java code is :
originateAction.setChannel(Local/[EMAIL PROTECTED]/n);
originateAction.setCallerId(asterisk);
originateAction.setCallingPres(new Boolean(true));
originateAction.setContext(mariaSIP);
originateAction.setExten(222);
originateAction.setPriority(nPriority);
originateAction.setTimeout(nTimeout);
originateResponse = managerConnection.sendAction(originateAction, 3);
if(originateResponse.getResponse().equals(Success))
{
setVarAction.setVariable(STRING3);
setVarAction.setValue(SIP/6020);
originateResponse = managerConnection.sendAction(setVarAction, 3);
if(originateResponse.getResponse().equals(Success))
{
RedirectAction redirectAction = new RedirectAction();
redirectAction.setChannel(sChannel);
redirectAction.setContext(mariaSIP);
redirectAction.setExten(444);
redirectAction.setPriority (new
Integer(1));
originateResponse = managerConnection.sendAction(redirectAction,
3); if (originateResponse.getResponse().equals(Success))
{
}
}
My problem is that in my manager console on 192.-...-.191 I dial, but
the other asterisk doesn´t seem to realiza I am dialing.. my console
says
Executing Wait(Local/[EMAIL PROTECTED],2, 1) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]:5038,20) in new stack
-- Called [EMAIL PROTECTED]:5038
-- Nobody picked up in 2 ms
-- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack
== Spawn extension (mariaSIP, 16007, 3) exited non-zero on
'Local/[EMAIL PROTECTED],2'
Can any body help me? I will be very gratefull for any help. Thanks in advance,
--
María
___
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[Asterisk-Users] res_features pickupexten
is where anyone who knows what is needed to get the pickupexten (*8) running ? gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff I've activated it in features.conf (default *8) and also tested other extensions res_features.so is loaded show features says: Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# #1 Attended Transfer *2 One Touch Monitor *1 Disconnect Call * *0 the callgroup/pickupgroup settings are correct... dialing *8 or *8# on any client (zap/sip/sccp) results in unknown extension... using the automon-feature with *1 does work ...or is this feature only possible in the cvs-tree ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Hipath interconnections
Hi Stephen,You said that PRI works great. We are using HiPath 3550 and Siemens digital phone which using *11, *97 etc for function keys. However Asterisk uses the the * key plus one or two digits for function keys as well(it is common key combination for functions). So is it any way to disable *11, *97 keys in HiPath system and pass this keys to Asterisk?Thanks and regards,IsaacHi VictorLooking for the same answers here too. We are regional distributors for Hicom HiPath in this part of the world and until now we are still waiting for chan_cornet to come around. So far we have successfully interconnected via BRI (mISDN) and PRI (Zaptel) and it works great.Let's see if it's too good to be true soon.Best regards,Stephen ___ --Bandwidth 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] Linksys WIP300 WiFi Phone
On 2/27/06, amna saleem [EMAIL PROTECTED] wrote: umm.. Can you please tell me what phone u r talking about??i mean does it support IAX. Actually i am sick and tired of my DIAX and want a new IAX phone... I am using an older version of * like 1.0.3 I hope u will not mind replying to me It is a SIP Wifi phone. It doesn't support IAX. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] BLF not working after reload
On 2/26/06, mustardman29 [EMAIL PROTECTED] wrote: According to this blurb I found on the Asterisk Wiki, it was supposed to be fixed so it still works after a reload. Your suggestion is all fine and dandy but does nothing to rectify a server reboot. If phones have to be rebooted everytime the Asterisk server is rebooted or the sip.conf is reloaded just to allow BLF to keep working then this is a show stopper for me! Update Aug. 2005 (for Asterisk 1.2.0) After months in the bug tracker (bug 3644), we've finally committed a lot of changes to the SIP Subscribe subsystem in Asterisk cvs head: It now works even if you reload the dial plan It does not accept subscriptions to extensions without hints It will terminate subscriptions if the hint does not exist after a dialplan reload To get this to work properly, you Add a hint to the dialplan for the extension Optional: Configure incominglimit for the device (renamed to call-limit in Asterisk v.1.2) Optional: Enable notifyringing = yes if you'd also like to see the RINGING state to be notified -Original Message- From: Douglas Garstang [mailto:[EMAIL PROTECTED] Sent: Sunday, February 26, 2006 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] BLF not working after reload If you do a 'reload' in Asterisk, it deletes all the sip subscriptions. Do a 'sip show subscriptions' before and after a reload command. They will disappear. I've been bitching about this for a while, and asking why subscriptions can't be stored in astdb like registrations. If you reboot the phone, it sends the SIP SUBSCRIBE message to Asterisk again, which remembers it until the next reload. If you reboot the Astrisk server, you obviously lose it as well, because Asterisk is storing them in memory (not astdb). One workaround, is to not issue 'reload' commands. Just reload the module you've changed. I think reloading SIP will delete the subscriptions. For example, if you change the dial plan just issue an 'extensions reload'. Your subscriptions should remain. Lets just hope it's a long time for you between alternations to sip.conf! See Bug #6047 pls. It's got a pointer to a branch of /trunk that does fix this with regard to subscriptions surviving a reload. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] mpg123 alternative?
- Original Message - From: Doug Lytle [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 23, 2006 9:33 PM Subject: Re: [Asterisk-Users] mpg123 alternative? Matt Roth wrote: We were using the rawplayer method on our server, but it ended up spawning hundreds of zombie processes. I talked to Kevin Fleming about it, and he recommended switching to native MOH. Scalability is a big concern of mine, so I asked him about the I've had issues with Native MOH when using IAX trunking and Placing a caller into a parking slot. Sound is awful. So, for parking I use mpg123 and everything else I'm using Native MOH. Have you placed a bug report about this? Chris ___ --Bandwidth 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] mpg123 alternative?
Chris Stenton wrote: I've had issues with Native MOH when using IAX trunking and Placing a caller into a parking slot. Sound is awful. So, for parking I use mpg123 and everything else I'm using Native MOH. Have you placed a bug report about this? No, I can't always reproduce it. Seems to happen randomly. But, happens enough that I get complaints. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom bootrom and SIP software
I know this shouldn't be the place to ask this, but I've just tried to upgrade my IP600 with bootrom 2.6.2 and SIP 1.5.2 and I'm getting intotrouble here (I chose not to go to the higher software levels since there's a warningabout using"secure" links.. I am not trying to change anything functionally but thishas been a long outstanding upgrade. Now when I did the upgrade I found no BootRom.ver The log files(one of them anyway) seem to indicate theproblemis with loading the BootRom.ld I'm a complete layman with this. I do not know what the bootrom.ld does, what the others do..but the phone is in a loop now (gives an error and tries to reboot). If this is somehow a known problemhit when upgrading, I should have seen it from the long history of messages I have. Any help would be appreciated. I could post things but I think it better to find someone who knows what they want to read first. BEGIN:VCARD VERSION:2.1 N:Tarzi;AbdelRahman el FN:AbdelRahman el Tarzi ORG:Arab Banking Corporation;Proprietary Investment TITLE:Structured Credit Derivatives NOTE;ENCODING=QUOTED-PRINTABLE:Fax: +973 39 33 27 69=0D=0AContacts in Egypt: =0D=0ACell: +20(10) 1236700= =0D=0ACairo: Residence: +20 (2) 4028860=0D=0AMarina: Residence: +20 (46) 406= 2197 (temp unavailable)=0D=0AZomorroda: Residence: +20 (3) 5210765=0D=0A TEL;WORK;VOICE:+973 1754 3700 TEL;HOME;VOICE:+973 17 69 80 24 TEL;CELL;VOICE:+973 39 68 57 00 TEL;WORK;FAX:+973 1753 1427 ADR;WORK:;3rd floor, ABC Building;P.O. BOX 5698;Manama;;;Bahrain LABEL;WORK;ENCODING=QUOTED-PRINTABLE:3rd floor, ABC Building=0D=0AP.O. BOX 5698=0D=0AManama=0D=0ABahrain ADR;HOME;ENCODING=QUOTED-PRINTABLE:;;House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain;Manama;;= ;Bahrain LABEL;HOME;ENCODING=QUOTED-PRINTABLE:House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain=0D=0AManam= a=0D=0ABahrain X-WAB-GENDER:2 URL;WORK:www.arabbanking.com BDAY:20050123 KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZWMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTEzNTFaFw0wNjExMTEwOTEz NTFaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAeWFob28u Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQCvGOn8FwM/UUm7OYMdFZYn+hUrmDYo ARJGJvFDu7lnbrT/v3tf1zRpOULT8yN2PXtSUmsxlvYX2SCJ8PggECGGbyJEkd8bHmPJEi7g FHNs9h3ps7SJ+gQFkqa0soxegfHgQzrjrOGXNI1dMCKaYc6a2dSWRUBj4C1ii1dHYs7jmQID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQHlhaG9vLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAC9Tm59BZjKmw61xcYa4yXhPSqfkXTJy6eAVX4LSwM1gkRbV6HWZ HjQBmEhTkfrAF01xeKrDRh6vJIYGjSuPJRVmCN2+BA/UuNnK3EQOI+mwuku8KQzDAFXpJHhe +J5626T7NiuADtT2F0L3tLoFf8vvLcyTzvCHU+y6E2Danaak KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZXMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTE5MDRaFw0wNjExMTEwOTE5 MDRaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAZ21haWwu Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQDASKRiH2YqhCqPF3HDlPCdtHZb78Pn Z4S/qzgdLVdzeE1b2Ddd4gl+FkQw2IS4Q+3XSwsGyh9wY6irNb+nIrr5Gs9+JmpQTSPjQp72 trLvD+PvFetwQMotRODVsgxHIpgcTFBjpMZ4P24NeAGRBNzfPjwqx3gfscd10fWtiXGo8wID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQGdtYWlsLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAAZ2rAEswRkNEgiMcy3enKlTcQ9QiIFeQP5bq7iXDUkbhtcZHDdi ol+HaN6QyO2ZUCYbuK1d12VD92QpZuRxw0lS7K7qWU7aF5gabpnEjl1KQ0ujr+gEcV2ogvZY 2F4SZ7H9uF0c06/NT5TpoFyok3wJ/jZXJhRAbR/Eye678OCq KEY;X509;ENCODING=BASE64: MIICfDCCAeWgAwIBAgIDD80vMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMDYyMTMzMzVaFw0wNjExMDYyMTMz MzVaMG8xDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxJTAjBgkqhkiG9w0BCQEWFmFydGFyemlAYmF0ZWxj by5jb20uYmgwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAK+koXkgs50JRrsTV4tj2QS7 uZ05+iKe/lhkdv56a6oEUcw4tO03rGMcB+ocWwfmmIbZ1n5p8dRjybsZMI5zEnRsf/KeQLl3 1wBPYoKzVDQrulNMGh8FmhK8uWsW1FZSKJkbxZWjcI2fkbDLmQuvWBUdlgiOFOLp08m9bMvf ZpCfAgMBAAGjMzAxMCEGA1UdEQQaMBiBFmFydGFyemlAYmF0ZWxjby5jb20uYmgwDAYDVR0T AQH/BAIwADANBgkqhkiG9w0BAQQFAAOBgQA/TNRreOLNx7d1f7H9vfrnlTRuftVHVL4f6h6X u2Od18TDDP6/iUuiTtcMQfOOwiBBxjkgdupsDi4q8FrOseWu5ylM9hNg+1mtjSQT00CL6n4A CIh94LiywiMeJmxzKLuihUxyQu2aRFksaQS4unmENCZ23a+xB4DHuTD9V3FcAx== EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;PREF;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] REV:20060227T131602Z END:VCARD ___ --Bandwidth
[Asterisk-Users] how to configure my [EMAIL PROTECTED] 1.0.9 to do call forwarding ?
Hello everyone
The PBX is connected using 4 Line FXO card to the PSTN.
I wish to send calls that come to extension X to an external phone
number, i.e. call the comes from Line1 would go out using Line{2,3,4}.
I wish the user that the extension belongs to him be able to set it.
Can this be done ?
Can it be done from the user's phone (Sipura 841) ?
Can I as the wwwadmin user set it ?
Thank you.
My knowledge in dialing rules is rather limited (i.e. null...)
--
Cheers,
Maxim Vexler (hq4ever).
Do u GNU ?
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[Asterisk-Users] Newbie h323 question
Greetings, Complete newbie question so apologies here. I am trying to connect our test Asterisk server with a number of SIP clients to a H323 PSTN gateway, the basic connection of SIP Asterisks works a treat however the h323 is causing problems. Box is a Cisco IP-IP gateway running in non-gatekeeper mode but I cannot work out what to configure to make this work. Sip Asterisk Cisco IP-IP is the call flow. I have this working with our call manager internally replacing the Asterisk server however we are trying to evaluate the IP-IP gateway so any assistance in this would be gratefully appreciated. Phil Phil Clarkson Internet Architect email: [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] BLF not working after reload
I solved that problem for Polycom phones with the patch at: http://bugs.digium.com/view.php?id=6047 ___ --Bandwidth 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 and Hipath interconnections
If Siemens claims it is Open source, they also should provide the download link for the software...otherwise it wouldn't be OPEN source - Original Message - From: Tele Cost Price Reducer To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, February 27, 2006 10:25 AM Subject: Re: [Asterisk-Users] Asterisk and Hipath interconnections hi all, maybe i am mistaken but it seems to me that the HiPath 2000 series is an Asterisk based system. why am i saying this? because Siemens announce it is a Linux, Open Source system. so, as i do not know any OTHER PBX Linux- Open Source system rather then Asterisk, does anybody know something else? otherwise, if it is an Asterisk system, so why there is a need for Cornet? you can interconnect with IAX, isn't it? Mickey On 2/27/06, Stephen Arulraj [EMAIL PROTECTED] wrote: Hi VictorLooking for the same answers here too. We are regional distributors forHicom HiPath in this part of the world and until now we are still waiting for chan_cornet to come around. So far we have successfullyinterconnected via BRI (mISDN) and PRI (Zaptel) and it works great.Let's see if it's too good to be true soon.Best regards,Stephen Viktor Tatianin wrote:HelloCan anyone know where may download chan_cornet for interconnection Asteriskand Hipath via IPThanksViktor___ --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 listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--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
[Asterisk-Users] SPA-941 Selective DND
Hello, I have a request from a customer that I'm not sure how to implement. They have a Snom-360 as receptionist phone and SPA-941 for all other phones. They use the SPA-941 DND function when they are away from their desks, which happens often due to the nature of their business. They would like to have the SPA-941 accept internal calls while DND is set. If any of you know how to make this happen, I'd very much appreciate your help. The paging feature is not what they want, and the SPA-041 ignores the answer-after=0 SIP header when DND is on anyway. Thanks much Darren Ellis ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with chan-capi: outgoing calls on two lines
Hello, while testing the following scenario, I ran into trouble: One * box with two AVM active controllers in Point-to-Point-Mode is connected to another * box with ZapHFC/Quad-BRI cards using bristuff in NT-mode. All is working fine, I can call from one box to the other and vice versa. But if I'll cut one line, it is not possible to place an outbound call from chan-capi accross the still existing line. In detail: When all lines are connected, the first two calls are placed on line 1 (which is on controller 1). The next two calls are placed on line 2 (on controller 2) If I'll cut line 2, all works as expected (I can place two calls on line 1). But if I'll cut line 1, leaving line 2 up and running, I can not place any call. The CLI tells something about Protocol error layer 1 (broken line or B-channel removed by signalling protocol) and No one is available to answer at this time (1:0/0/0) If I do the same thing in the opposite direction (Calls are initiated from the other box with bristuff in NT-mode), all works fine. What am I doing wrong (or is this a bug)? Thanks in advance, Karsten ___ --Bandwidth 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] SPA-941 Selective DND
On 2/27/06, Darren Ellis [EMAIL PROTECTED] wrote: Hello, I have a request from a customer that I'm not sure how to implement. They have a Snom-360 as receptionist phone and SPA-941 for all other phones. They use the SPA-941 DND function when they are away from their desks, which happens often due to the nature of their business. They would like to have the SPA-941 accept internal calls while DND is set. If any of you know how to make this happen, I'd very much appreciate your help. The paging feature is not what they want, and the SPA-041 ignores the answer-after=0 SIP header when DND is on anyway. IF the SPA-941 doesn't support the selective DND feature, the only solution that comes to mind is to use server-side DNDs. ie. *11 (for example) to turn DND on and *12 to turn it off. In Asterisk, configure *11 to set a DND variable (database put DND ext yes/no). When somebody calls the extension, if DND is yes in the Asterisk internal DB for the extension, and the call is from a local channel, ring the SPA-941, otherwise send to voicemail. Make sense? Hope this helps. Regards, Gonzalo. ___ --Bandwidth 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] jitterbuffer and DTMF conflict?
I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaround (other than jitterbuffer=off)? - Mike ___ --Bandwidth 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] Asttapi - what's wrong?
When I try to call from asttapi one number, I get message No one is available to answer at this time (1:0/0/0). Immediately after that I try to call the same number from SIP phone (the same one that is used with asttapi) and call goes true. What have I done wrong? This is how it looks on CLI. == Parsing '/etc/asterisk/manager.conf': Found == Manager 'tomo' logged on from 10.0.0.203 Channel SIP/341-062e was answered. -- Executing Dial(SIP/341-062e, OOH323/[EMAIL PROTECTED]) in new s tack -- Called [EMAIL PROTECTED] == No one is available to answer at this time (1:0/0/0) -- Executing Hangup(SIP/341-062e, ) in new stack == Spawn extension (sip, 00989970434, 2) exited non-zero on 'SIP/341-062e' == Manager 'tomo' logged off from 10.0.0.203 -- Executing Dial(SIP/341-9e85, OOH323/[EMAIL PROTECTED]) in new s tack -- Called [EMAIL PROTECTED] -- OOH323/85.114.35.42-b1b4 is ringing == Spawn extension (sip, 00989970434, 1) exited non-zero on 'SIP/341-9e85' ___ --Bandwidth 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] jitterbuffer and DTMF conflict?
I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaround (other than jitterbuffer=off)? Might try turning off trunking (assuming you have it turned on) and test again. Seems a couple of parameters interact and probably has something to do with different versions of iax. ___ --Bandwidth 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] courtesy message calling mobile phones
Hi everybody. Just noticed that when calling a mobile phone, Asterisk doesnt bridge the voice message by telco if mobile is unreachable, but keeps on ringing till it receives a hangup signal. I think this is due to the fact that the message is played without the call has been answered, but Im wondering if theres some way to let Asterisk realize it. All I see in the CLI is the line PROGRESS with cause code 0 received. Thank you, _fangi_ ___ --Bandwidth 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] jitterbuffer and DTMF conflict?
Rich Adamson wrote: I find that DTMF does not work reliably if jitterbuffer=on for certain Can anyone suggest a workaround (other than jitterbuffer=off)? Might try turning off trunking (assuming you have it turned on) and test again. Seems a couple of parameters interact and probably has something to do with different versions of iax. Rich, I'm not sure if trunking is on by default, but I turned it off explicitly. No difference, sadly. - Mike ___ --Bandwidth 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] Problem with chan-capi: outgoing calls on two lines
On Mon, 27 Feb 2006, Karsten Wemheuer wrote: while testing the following scenario, I ran into trouble: One * box with two AVM active controllers in Point-to-Point-Mode is connected to another * box with ZapHFC/Quad-BRI cards using bristuff in NT-mode. All is working fine, I can call from one box to the other and vice versa. But if I'll cut one line, it is not possible to place an outbound call from chan-capi accross the still existing line. In detail: When all lines are connected, the first two calls are placed on line 1 (which is on controller 1). The next two calls are placed on line 2 (on controller 2) If I'll cut line 2, all works as expected (I can place two calls on line 1). But if I'll cut line 1, leaving line 2 up and running, I can not place any call. The CLI tells something about Protocol error layer 1 (broken line or B-channel removed by signalling protocol) and No one is available to answer at this time (1:0/0/0) If I do the same thing in the opposite direction (Calls are initiated from the other box with bristuff in NT-mode), all works fine. What am I doing wrong (or is this a bug)? This is not a bug, just normal behaviour. chan_capi does not know about the status of the ISDN line, it assumes to be usable when configured. So when you try to dial out chan_capi will choose a channel/line according to internal list of free channels and selects it in the CAPI request. When the driver reports an error via CAPI, chan_capi just signals this error to Asterisk. There is no logic in chan_capi to do something like: If the controller 1 isn't ready, use controller 2. The same happens if the b-channels are already used by another application/device. Armin ___ --Bandwidth 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] chan iax2 auto congest
Hello, sometimes I'm experiencing autocongest error due slow response, anyone knows, what this means? Second or third attempt after that happens pass successfully... this happens ever in fastethernet lan, so no problem with lag in wan environment, I'm using idefisk 1.32 on client side (winxp or linux)... PJ -- Executing Dial(IAX2/bill-7, IAX2/963) in new stack -- Called 963 Feb 27 15:59:32 NOTICE[6283]: chan_iax2.c:2821 auto_congest: Auto-congesting call due to slow response -- IAX2/963-18 is circuit-busy -- Hungup 'IAX2/963-18' == Everyone is busy/congested at this time (2:0/1/1) ___ --Bandwidth 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] Problems dialing to another Asterisk server
At first sight there is no problem, your code looks good, no warnings
etc, is just that nobody picks up in the other end. Do you have access
to the other Asterisk server? what does the console shows up? I have
not used the manager with Java, but that does not seems to be your
problem. I guess you should try to isolate the problem. Delete all the
manager and java things, and attempt to make a direct dial with some
phone to SIP/6021 blah
RegardsOn 2/27/06, María Chóliz [EMAIL PROTECTED] wrote:
Hi,I have a problem dialing a SIP phone which is logged in as differentAstesrik machine from the one I am working with.I want to call a phone in Another astersik machine in , if it answers,calling a SiP phone registered in my ASterisk:
My dialplan is:[mariaSIP]exten = _1.,1,Wait(1)exten = _1.,2,Dial(SIP/[EMAIL PROTECTED]:5038,20)exten = _1.,3,HangUp()exten = 222,1,MusicOnHold()exten = 444,1,Dial(${STRING3})
exten = 444,2,RingingSIP/6021 is the telephone logged in the another machine, which is192.168.0.51 and asterisk is listening in 192.168.0.51
, port 5038And my Manager-java code is :originateAction.setChannel(Local/[EMAIL PROTECTED]/n);originateAction.setCallerId(asterisk);originateAction.setCallingPres(new Boolean(true));
originateAction.setContext(mariaSIP);originateAction.setExten(222);originateAction.setPriority(nPriority);originateAction.setTimeout(nTimeout);originateResponse = managerConnection.sendAction
(originateAction, 3);if(originateResponse.getResponse().equals(Success)){ setVarAction.setVariable(STRING3); setVarAction.setValue(SIP/6020); originateResponse =
managerConnection.sendAction(setVarAction, 3); if(originateResponse.getResponse().equals(Success)){RedirectAction redirectAction = new RedirectAction();redirectAction.setChannel
(sChannel);redirectAction.setContext(mariaSIP);redirectAction.setExten(444);redirectAction.setPriority (newInteger(1));originateResponse = managerConnection.sendAction
(redirectAction,3);if (originateResponse.getResponse().equals(Success)){} }My problem is that in my manager console on 192.-...-.191 I dial, but
the other asterisk doesn´t seem to realiza I am dialing.. my consolesaysExecuting Wait(Local/[EMAIL PROTECTED],2, 1) in new stack-- Executing Dial(Local/[EMAIL PROTECTED]
,2,SIP/[EMAIL PROTECTED]:5038,20) in new stack-- Called [EMAIL PROTECTED]:5038-- Nobody picked up in 2 ms-- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack
== Spawn extension (mariaSIP, 16007, 3) exited non-zero on'Local/[EMAIL PROTECTED],2'Can any body help me? I will be very gratefull for any help. Thanks in advance,--María___
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org
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RE: [Asterisk-Users] BLF not working after reload
Marco, Which versions of Asterisk will that patch work with? Douglas. -Original Message- From: Marco Maiolini [mailto:[EMAIL PROTECTED] Sent: Monday, February 27, 2006 6:36 AM To: asterisk-users Subject: Re: [Asterisk-Users] BLF not working after reload I solved that problem for Polycom phones with the patch at: http://bugs.digium.com/view.php?id=6047 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Hipath interconnections
Hi You may write trunk withdial digits *11 which connect to asterisk via PRI or BRI Viktor -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Isaac XiaoSent: Monday, February 27, 2006 1:44 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk and Hipath interconnections Hi Stephen,You said that PRI works great. We are using HiPath 3550 and Siemens digital phone which using *11, *97 etc for function keys. However Asterisk uses the the * key plus one or two digits for function keys as well(it is common key combination for functions). So is it any way to disable *11, *97 keys in HiPath system and pass this keys to Asterisk?Thanks and regards,IsaacHi VictorLooking for the same answers here too. We are regional distributors for Hicom HiPath in this part of the world and until now we are still waiting for chan_cornet to come around. So far we have successfully interconnected via BRI (mISDN) and PRI (Zaptel) and it works great.Let's see if it's too good to be true soon.Best regards,Stephen ___ --Bandwidth 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] TE411P problem-- probably stupid.
Hi, Probably a stupid syntax problem.. but can't seem to make these files work for more than the first T1 (ok, if I comment out the 2nd, 3rd, and 4th span). Can someone proofread this for me? Thanks! == zaptel.conf span=1,0,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 span=3,0,0,esf,b8zs bchan=49-71 dchan=72 span=4,0,0,esf,b8zs bchan=73-95 dchan=96 loadzone=us defaultzone=us zapata.conf ; ; Zapata telephony interface ; ; Configuration file [channels] busydetect=1 busycount=7 relaxdtmf=yes callerid=asreceived context=default signalling=pri_cpe usecallerid=yes transfer=yes echocancel=yes echocancelwhenbridged=yes switchtype=dms100 rxgain=0.0 txgain=0.0 immediate=no signalling=pri_cpe group=1 context=default channel = 1-23 busydetect=1 busycount=7 relaxdtmf=yes callerid=asreceived context=default signalling=pri_cpe usecallerid=yes transfer=yes echocancel=yes echocancelwhenbridged=yes switchtype=dms100 rxgain=0.0 txgain=0.0 immediate=no signalling=pri_cpe group=2 context=default channel =25-47 busydetect=1 busycount=7 relaxdtmf=yes callerid=asreceived context=default signalling=pri_cpe usecallerid=yes transfer=yes echocancel=yes echocancelwhenbridged=yes switchtype=dms100 rxgain=0.0 txgain=0.0 immediate=no signalling=pri_cpe group=3 context=default channel = 49-71 busydetect=1 busycount=7 relaxdtmf=yes callerid=asreceived context=default signalling=pri_cpe usecallerid=yes transfer=yes echocancel=yes echocancelwhenbridged=yes switchtype=dms100 rxgain=0.0 txgain=0.0 immediate=no signalling=pri_cpe group=4 context=default channel = 73-95 == ___ --Bandwidth 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 Default Ring Volume
On 2/25/06, Anton Krall [EMAIL PROTECTED] wrote: Does anybody know how to set polycom's default ring volume ? Everytime you restart a polycom phone, ring defaults to a very low volume setting which is kind of annoying having to set everytime you reboot. IIRC, You have to set it in the XML file and reprovision automatically each time the phone reboots. ___ --Bandwidth 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: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
I have also issues with jitter over wan (cdma), I'm trying to debug how dejitter buffer is working (using iax2 jb debug), but nothing happens/no debug output on asterisk console :-( is any way how to monitor iax jitter buffer? thx PJ ___ --Bandwidth 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] chanspy instability
Teehee... no I didn't do any of that.. mostly because it's a feature I don't use all that often, and at the moment I can't upgrade :) so... On 2/24/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Matt wrote: I too have noticed this but received no solution =\ I was running 1.2.0 Did you try it again after updating to the latest 1.2 release? Did you report the bug on the bug tracker and provide a backtrace so someone could try to fix it? If not, how did you expect a solution to be created? We aren't telepathic, you know :-) ___ --Bandwidth 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] BLF not working after reload
Thanks for pointing that bug out to me BJ. At least I understand what is going on now. It's definitely not limited to the Polycom Phones. -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Monday, February 27, 2006 3:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BLF not working after reload On 2/26/06, mustardman29 [EMAIL PROTECTED] wrote: According to this blurb I found on the Asterisk Wiki, it was supposed to be fixed so it still works after a reload. Your suggestion is all fine and dandy but does nothing to rectify a server reboot. If phones have to be rebooted everytime the Asterisk server is rebooted or the sip.conf is reloaded just to allow BLF to keep working then this is a show stopper for me! Update Aug. 2005 (for Asterisk 1.2.0) After months in the bug tracker (bug 3644), we've finally committed a lot of changes to the SIP Subscribe subsystem in Asterisk cvs head: It now works even if you reload the dial plan It does not accept subscriptions to extensions without hints It will terminate subscriptions if the hint does not exist after a dialplan reload To get this to work properly, you Add a hint to the dialplan for the extension Optional: Configure incominglimit for the device (renamed to call-limit in Asterisk v.1.2) Optional: Enable notifyringing = yes if you'd also like to see the RINGING state to be notified -Original Message- From: Douglas Garstang [mailto:[EMAIL PROTECTED] Sent: Sunday, February 26, 2006 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] BLF not working after reload If you do a 'reload' in Asterisk, it deletes all the sip subscriptions. Do a 'sip show subscriptions' before and after a reload command. They will disappear. I've been bitching about this for a while, and asking why subscriptions can't be stored in astdb like registrations. If you reboot the phone, it sends the SIP SUBSCRIBE message to Asterisk again, which remembers it until the next reload. If you reboot the Astrisk server, you obviously lose it as well, because Asterisk is storing them in memory (not astdb). One workaround, is to not issue 'reload' commands. Just reload the module you've changed. I think reloading SIP will delete the subscriptions. For example, if you change the dial plan just issue an 'extensions reload'. Your subscriptions should remain. Lets just hope it's a long time for you between alternations to sip.conf! See Bug #6047 pls. It's got a pointer to a branch of /trunk that does fix this with regard to subscriptions surviving a reload. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] courtesy message calling mobile phones
Can you explain this? What country? In this case it's not asterisk but the telco that has to do the Answer. To every mobile? or just that provider? On 2/27/06, Francesco Angi [EMAIL PROTECTED] wrote: Hi everybody. Just noticed that when calling a mobile phone, Asterisk doesn't bridge the voice message by telco if mobile is unreachable, but keeps on ringing till it receives a hangup signal. I think this is due to the fact that the message is played without the call has been answered, but I'm wondering if there's some way to let Asterisk realize it. All I see in the CLI is the line PROGRESS with cause code 0 received. Thank you, _fangi_ ___ --Bandwidth 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] SPA-941 Selective DND
So why use DND? As far as the phone knows, they are all internal/external. You should realy look into an asterisk side extension that will block incoming calls. On 2/27/06, Darren Ellis [EMAIL PROTECTED] wrote: Hello, I have a request from a customer that I'm not sure how to implement. They have a Snom-360 as receptionist phone and SPA-941 for all other phones. They use the SPA-941 DND function when they are away from their desks, which happens often due to the nature of their business. They would like to have the SPA-941 accept internal calls while DND is set. If any of you know how to make this happen, I'd very much appreciate your help. The paging feature is not what they want, and the SPA-041 ignores the answer-after=0 SIP header when DND is on anyway. Thanks much Darren Ellis ___ --Bandwidth 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] automon not working for analogue phone
Hi all, I have just setup automon functionality on my asterisk box and when trying to activate the feature by pressing *1 on my analogue phone within the conversation it does not work. That is strange because with SIP phone it works OK. Does anyone know what could be wrong? Thanks, Ondrej ___ --Bandwidth 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: Rewind MusicOnHold?
Hi guys, Matt gave the advice belowfora way to cause MoH to rewind and play from the beginning for each call that comes in. However the music doesn't restart. Here was my first attempt:- mode=customdirectory=/var/lib/asterisk/mohmp3application=/usr/local/bin/madplay -Q -o raw:- --mono -R 8000 -a -12 So i added Time and Repeat to the command line, but that still didnt work: mode=customdirectory=/var/lib/asterisk/mohmp3application=/usr/local/bin/madplay -Q -o raw:- --mono -R 8000 -a -12--time=00:00:60.000 --repeat=1 Does anyone have any ideas on how I should be doing this? Thanks Dan Journo www.TextOver.com On 03/02/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: Dan Journo wrote: Ok, i feel like im getting somewhere but i need a little help. Asterisk displays this when its loading:- [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' == Parsing '/etc/asterisk/musiconhold.conf': FoundUse a custom musiconhold class playing ulaw files or whatever - they will start from the beginning each time.--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)___--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] automon not working for analogue phone
Try pressing FLASH, then *1 and then FLASH again. Michelle Dupuis Technical Support Specialist Oxford Consulting Group Ltd. Making IT work for your business... T: (519) 672-8238 E: [EMAIL PROTECTED] W: www.ocg.ca -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ondrej Valousek Sent: Monday, February 27, 2006 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] automon not working for analogue phone Hi all, I have just setup automon functionality on my asterisk box and when trying to activate the feature by pressing *1 on my analogue phone within the conversation it does not work. That is strange because with SIP phone it works OK. Does anyone know what could be wrong? Thanks, Ondrej ___ --Bandwidth 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] SPA-941 Selective DND
We are working on a smartDND agi script which will do this. Should be coming out this spring :) MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, February 27, 2006 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-941 Selective DND So why use DND? As far as the phone knows, they are all internal/external. You should realy look into an asterisk side extension that will block incoming calls. On 2/27/06, Darren Ellis [EMAIL PROTECTED] wrote: Hello, I have a request from a customer that I'm not sure how to implement. They have a Snom-360 as receptionist phone and SPA-941 for all other phones. They use the SPA-941 DND function when they are away from their desks, which happens often due to the nature of their business. They would like to have the SPA-941 accept internal calls while DND is set. If any of you know how to make this happen, I'd very much appreciate your help. The paging feature is not what they want, and the SPA-041 ignores the answer-after=0 SIP header when DND is on anyway. Thanks much Darren Ellis ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] billing - different tarif per phone
Hello, I would like apply different call rate (tarif) per outgoing number (or group of phones, based on prefixes), I'm playing with astpp, but seems, that this feature isn't available here, can you recommend any other open-source billing (A2billing, AstBill?), that this can do? thank you! PJ ___ --Bandwidth 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: Rewind MusicOnHold?
As a follow on from my last email, it appears that Asterisk restarts the player application if the process terminates. Does anyone know a way to stop that? Thanks Dan On 27/02/06, Dan Journo [EMAIL PROTECTED] wrote: Hi guys, Matt gave the advice belowfora way to cause MoH to rewind and play from the beginning for each call that comes in. However the music doesn't restart. Here was my first attempt:- mode=customdirectory=/var/lib/asterisk/mohmp3application=/usr/local/bin/madplay -Q -o raw:- --mono -R 8000 -a -12 So i added Time and Repeat to the command line, but that still didnt work: mode=customdirectory=/var/lib/asterisk/mohmp3application=/usr/local/bin/madplay -Q -o raw:- --mono -R 8000 -a -12--time=00:00:60.000 --repeat=1 Does anyone have any ideas on how I should be doing this? Thanks Dan Journo www.TextOver.com On 03/02/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: Dan Journo wrote: Ok, i feel like im getting somewhere but i need a little help. Asterisk displays this when its loading:- [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' == Parsing '/etc/asterisk/musiconhold.conf': FoundUse a custom musiconhold class playing ulaw files or whatever - they will start from the beginning each time.--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)___ --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] Polycom Default Ring Volume
Yep, that much I know but do you know which setting to use? Manual doesn't mention anything. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Wilson Pickett |Sent: Monday, February 27, 2006 10:12 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Polycom Default Ring Volume | |On 2/25/06, Anton Krall [EMAIL PROTECTED] wrote: | Does anybody know how to set polycom's default ring volume ? |Everytime | you restart a polycom phone, ring defaults to a very low volume | setting which is kind of annoying having to set everytime you reboot. | |IIRC, You have to set it in the XML file and reprovision |automatically each time the phone reboots. |___ |--Bandwidth 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] Problem with chan-capi: outgoing calls on two lines
Hello, On Mo, 27 Feb 2006, Armin Schindler wrote: On Mon, 27 Feb 2006, Karsten Wemheuer wrote: In detail: When all lines are connected, the first two calls are placed on line 1 (which is on controller 1). The next two calls are placed on line 2 (on controller 2) If I'll cut line 2, all works as expected (I can place two calls on line 1). But if I'll cut line 1, leaving line 2 up and running, I can not place any call. The CLI tells something about Protocol error layer 1 (broken line or B-channel removed by signalling protocol) and No one is available to answer at this time (1:0/0/0) If I do the same thing in the opposite direction (Calls are initiated from the other box with bristuff in NT-mode), all works fine. What am I doing wrong (or is this a bug)? This is not a bug, just normal behaviour. chan_capi does not know about the status of the ISDN line, it assumes to be usable when configured. So when you try to dial out chan_capi will choose a channel/line according to internal list of free channels and selects it in the CAPI request. When the driver reports an error via CAPI, chan_capi just signals this error to Asterisk. There is no logic in chan_capi to do something like: If the controller 1 isn't ready, use controller 2. Ok, I didn't know the details of the capi layer. The same happens if the b-channels are already used by another application/device. ... which would not happen on a line in point-to-point mode. And I just checked, that all works ok, if the channels are in use of asterisk itself (incomming calls). So all is ok, except that there seems to be no possibility to check the lower layer state e.g. a broken line. (In case of zap You can look at the files in /proc/zap). Thanks for Your quick response Karsten ___ --Bandwidth 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] courtesy message calling mobile phones
C F wrote: Can you explain this? What country? In this case it's not asterisk but the telco that has to do the Answer. To every mobile? or just that provider? I too have seen something similar in the past. When calling Verizon (408-489) numbers, when there is no answer and it rolls over to voicemail, the callee's greeting plays with no answer indication from Verizon. Eventually the Dial times out while in the middle of the callee's greeting and the caller is not able to leave a voicemail. I don't know if this is still happening now or if it was a just a temporary fluke when it was reported to me. -Johnathan ___ --Bandwidth 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] jitterbuffer and DTMF conflict?
On Feb 27, 2006, at 6:09 AM, Dr. Michael J. Chudobiak wrote: I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaround (other than jitterbuffer=off)? Actually I don't think Asterisk should jitter buffer in the above case? I am a newb, so be warned. My research seems to indicate that jitter buffering should only be used at the end points, as that is where the audio needs to be reassembled. Since in this case asterisk is the man in the middle and not one of the endpoints (I think?) it doesn't need to jitter buffer at all for calls being placed through an outside IAX carrier? If what I have written is true, then jitter buffering is only adding extra latency. If you are using Zap channels the above is probably wrong though? I noticed also, from one of my handsets attached to an ATA (AG168v) connecting through IAX2, DTMF was sensitive to volume adjustment even though it is out of band (rfc2833). Another thing that might help in the case you describe is to use a more band width efficient codec like G729 or GSM versus uLaw or alaw. Sorry if this is all old hat to you and I am restating the obvious. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P digium card
Okay everyone Im moving away from using sipura 841 phones. Im starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to upgrade the TDM card and the phones? Basically, my users say the phone system is unusable as is. The sound quality is choppy and they cant understand people speaking on the other end. I dont want to swap out their IP phones and then find out they are seeing the same issues with that. Any help as always is greatly appreciated everyone. Thanks ! Nora Lavelle ___ --Bandwidth 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] Music on hold and conferencing on OS X
On Feb 26, 2006, at 8:03 PM, Joseph Blake wrote: We're setting up asterisk at the office (really doing some testing right now) and it is going to be hosted on a dual G5 XServe running OS X. I love it. Glad to hear it. Should be a monster. We're an apple certified solutions provider, etc. so we want to build all our stuff on apple hardware and software. Anyway, the last sticking point is moh and meetme. Is there any solution to get moh and meetme working on OS X? Meetme isn't necessarily a big deal for us in our setup, but we plan to start selling asterisk solutions to our customers and they might need/want a conference solution. Also, something that is somewhat of a big deal (but not a deal breaker) for us is music on hold. Is there any way to get moh working without zaptel drivers, or is there another timing source that asterisk can use that works on OS X? I certainly don't see why not. I am also a hardcore Apple guy, and a newb to asterisk. I have been using it for several months now and have a small setup running on a measly old G3/400 imac (with a dead screen). I will try to play with MOH, but I suggest that if you have more time for this, try using the newer asterisk's native music on hold method. I see no reason why this wouldn't work as it is playing back audio just like any other audio (which works fine). This means no MPG123. If this makes no sense to you try searching this list for mpg123 alternative. Let us know.. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to query a table from the keypad?
Thanks Chris and Mike for the great ideas.Richard"Chris A. Icide" [EMAIL PROTECTED] wrote: Or you could skip the overhead associated with an AGI and use thedialplan command availabe after installing asterisk-addons MYSQL.exten = _X.,1,Read(PO-NUMBER,enter-yr-po-num)exten = _X.,2,MYSQL(Connect connid)exten = _X.,3,MYSQL(Query resultid ${connid} SELECT balance FROMaccount-payables WHERE po_num=${PO-NUMBER})exten = _X.,4,MYSQL(Fetch fetch ${resultid} AMOUNT-DUE)exten = _X.,5,MYSQL(Clear ${resultid})Of course you will want to put in place all the error traps and whenusing this function I always have a check in my hangup routine to makesure I close the open mysql connection. So at the end of the abovedialplan, you should
have
the value you want in the AMOUNT-DUE variable.-ChrisMike Pollitt wrote: Hi Richard What you want is AGI: http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI You could write a perl script to read the PO number from stdin and spit back the balance (or whatever). To read the PO number from the user, use the Read() dialplan application. To play back the balance, you could use SayDigits() (but theres probably a more elegant solution specifically for speaking amounts of money). *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Richard Reina *Sent:* Friday, 24 February 2006 9:34 AM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] How to query a table
from the
keypad? I am trying to give users the option to query our accts. payable database by supplying their PO number. I able to write queries via perl-DBI-mysql but have no idea how to get * to do it from the IVR. Is this possible? Can anyone point me in the right direction for help or examples? Thanks, Richard What are the most popular cars? Find out at Yahoo! Autos ___ --Bandwidth and Colocation provided by Easynews.com -- Asteri
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Re: [Asterisk-Users] Internal Server Error
Im starting to get a lot of errores from asterisk when transfering calls from one phone to another: Incoming call: Got SIP response 500 Internal Server Error back from What does this error usually mean? I have exactly the same problem with Polycom phones while transferring... Strange... B. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RES: RTP and Signalling
Hi, I need to send RTP from asterisk to one IP and signalling to another IP. In this case, can you help me to arrange that configuration on sip.conf [] type=friend username= secret= host= dtmfmode=rfc2833 disallow=all allow=g729 Atenciosamente Diretoria Comercial - Newton Medina PABX 11.3898.0112 Fax 11.38980112 MSN[EMAIL PROTECTED] Rua Augusta 2.212 SL 26 Jardins 01412001 São Paulo - Brasil ___ --Bandwidth 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] Matching '*'
I'm trying to find a way in extensions.conf to match ANYTHING dialled, including characters such as *. The following works for numbers... exten = _X.,1,AGI(script) but doesn't catch when someone dialls * first. I tried this: exten = _.,1,AGI(script) which catches when someone dials say, *123 for example, but after the AGI script terminates, Asterisk executes it again with the 'h' extension. So then I tried... exten = _.,1,AGI(script) exten = _.,2,Hangup() which doesn't work. So, what exten regex can I use that would catch anything dialled, or how can I stop Asterisk from executing the AGI script a second time when I use "_."? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX provider recommendation wanted
On Feb 27, 2006, at 2:55 AM, Dr. Michael J. Chudobiak wrote: Can someone recommend an IAX provider for US DIDs who will: snip 3) Have great audio quality This is somewhat a meaningless question, as the route from you to the call terminating service can make or break the quality. You have to scrutinize the route from your ISP to the termination service in addition to finding one that meets your requirements on paper. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: courtesy message calling mobile phones
[EMAIL PROTECTED] is believed to have said: Can you explain this? What country? In this case it's not asterisk but the telco that has to do the Answer. To every mobile? or just that provider? Well, it's funny because here, now (Italy; Telecom Italia PSTN calling Wind mobile), I do get the courtesy message saying that they're moving me to voicemail, if I call myself from the office PBX to my mobile Wind number, and the cellphone is switched off. The question I would like to know more about is if there is some way to discriminate between a courtesy message and a carbon-based voice responder (aka 'a person' ;-). Or is there some way to match a prerecorded voice file to the audio stream coming in? This could be great to do -easily- something like the voice commands setups most cellphones offer: you say 'home', the voice file is matched, and you get the call going. Regards, Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P digium card
Nora, If you have issues with choppy calls, most likely your issue isnt with your phones or TDM400, but it sounds like you have some issues with your voip trunks and/or network connectivity issues. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nora Lavelle Sent: Monday, February 27, 2006 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TDM400P digium card Okay everyone Im moving away from using sipura 841 phones. Im starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to upgrade the TDM card and the phones? Basically, my users say the phone system is unusable as is. The sound quality is choppy and they cant understand people speaking on the other end. I dont want to swap out their IP phones and then find out they are seeing the same issues with that. Any help as always is greatly appreciated everyone. Thanks ! Nora Lavelle CONFIDENTIALITY NOTICE: This email and any attachments are intended only for the designated recipients. Superior IT Solutions prohibits use, distribution or transmittal by or to an inintended recipient without Superior IT Solution's express written approval. If you are not the intended recipient, please delete this email and notify Superior IT Solutions ([EMAIL PROTECTED])This message (And any attachment) has been scanned by F-Secure and NortonAnti-Virus before leaving our mail server.- ___ --Bandwidth 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] Problem with chan-capi: outgoing calls on two lines
On Mon, 27 Feb 2006, Karsten Wemheuer wrote: Hello, On Mo, 27 Feb 2006, Armin Schindler wrote: On Mon, 27 Feb 2006, Karsten Wemheuer wrote: In detail: When all lines are connected, the first two calls are placed on line 1 (which is on controller 1). The next two calls are placed on line 2 (on controller 2) If I'll cut line 2, all works as expected (I can place two calls on line 1). But if I'll cut line 1, leaving line 2 up and running, I can not place any call. The CLI tells something about Protocol error layer 1 (broken line or B-channel removed by signalling protocol) and No one is available to answer at this time (1:0/0/0) If I do the same thing in the opposite direction (Calls are initiated from the other box with bristuff in NT-mode), all works fine. What am I doing wrong (or is this a bug)? This is not a bug, just normal behaviour. chan_capi does not know about the status of the ISDN line, it assumes to be usable when configured. So when you try to dial out chan_capi will choose a channel/line according to internal list of free channels and selects it in the CAPI request. When the driver reports an error via CAPI, chan_capi just signals this error to Asterisk. There is no logic in chan_capi to do something like: If the controller 1 isn't ready, use controller 2. Ok, I didn't know the details of the capi layer. The same happens if the b-channels are already used by another application/device. ... which would not happen on a line in point-to-point mode. No, the line mode doesn't matter. If you have another application running using CAPI, chan_capi also doesn't know about the usage. And I just checked, that all works ok, if the channels are in use of asterisk itself (incomming calls). So all is ok, except that there seems to be no possibility to check the lower layer state e.g. a broken line. (In case of zap You can look at the files in /proc/zap). It would be possible to evaluate the error code and then try again with another channel, but that logic is just not implemented. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P digium card
I'm moving away from using sipura 841 phones. I'm starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to upgrade the TDM card and the phones? Basically, my users say the phone system is unusable as is. The sound quality is choppy and they can't understand people speaking on the other end. I don't want to swap out their IP phones and then find out they are seeing the same issues with that. We use IP501's with dual TDM400P's (6 x FXO) here and so far, everyone has been satisfied with performance. No big complaints other than occasional minor echo but nothing's a show stopper. Voice quality has been rated as very good rather than great as with our old Avaya PBX but for the price, I can't complain too much. I don't even have QoS setup yet so I expect performance to be even better once I get a new switch in. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P digium card
Thanks dewey. Any feedback on how to debug this issue ? -nora From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dewey Straughn Sent: Monday, February 27, 2006 11:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] TDM400P digium card Nora, If you have issues with choppy calls, most likely your issue isnt with your phones or TDM400, but it sounds like you have some issues with your voip trunks and/or network connectivity issues. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nora Lavelle Sent: Monday, February 27, 2006 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TDM400P digium card Okay everyone Im moving away from using sipura 841 phones. Im starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to upgrade the TDM card and the phones? Basically, my users say the phone system is unusable as is. The sound quality is choppy and they cant understand people speaking on the other end. I dont want to swap out their IP phones and then find out they are seeing the same issues with that. Any help as always is greatly appreciated everyone. Thanks ! Nora Lavelle CONFIDENTIALITY NOTICE: This email and any attachments are intended only for the designated recipients. Superior IT Solutions prohibits use, distribution or transmittal by or to an inintended recipient without Superior IT Solution's express written approval. If you are not the intended recipient, please delete this email and notify Superior IT Solutions ([EMAIL PROTECTED]) This message (And any attachment) has been scanned by F-Secure and Norton Anti-Virus before leaving our mail server. - ___ --Bandwidth 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] MWI
I am using an external voice mail system. I'd like to be able to light the message waiting light on SIP and SCCP phones. Can someone point me in the right direction? Is there a manager command or and AGI app that does this. If not, what would I have to do to interface with * and have the MWI light work? Wesley A. SchochetSenior Telecommunications EngineerSelect Comfort Corporation(763-551-7757651-592-5441*[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] IAX provider recommendation wanted
Martin Joseph wrote: snip 3) Have great audio quality This is somewhat a meaningless question, as the route from you to the call terminating service can make or break the quality. Sure, but some carriers have problems inside their own networks. I can optimize the routing to the provider as needed, but it doesn't matter if they aren't actively addressing support issues and their own connections. - Mike ___ --Bandwidth 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] Matching '*'
Douglas Garstang schrieb: I'm trying to find a way in extensions.conf to match ANYTHING dialled, Hi, your subject is probably not correct. You want to catch anything except h, t, ...? Maybe you want to get matched the digits and *. Thus try: _[*0-9]. This will match any dialed string, which starts with * or a digit and has at least a length of 2. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with HT 488 FXO
Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug: -- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK669516e2;rport From: Unknown sip:[EMAIL PROTECTED];tag=as073738f8 To: sip:192.168.1.157:5062;tag=ebc4a8e2 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Grandstream HT488 1.0.2.16 Contact: sip:[EMAIL PROTECTED]:5062;user=phone Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' asterisk1*CLI -- SIP read from 192.168.1.157:5062: SIP/2.0 481 No Such Call Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK669516e2;rport From: Unknown sip:[EMAIL PROTECTED];tag=as073738f8 To: sip:192.168.1.157:5062;tag=52242a6b Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Grandstream HT488 1.0.2.16 Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.1.157:5060: REGISTER sip:192.168.1.157 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK42700737;rport From: sip:[EMAIL PROTECTED];tag=as558874a4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 120 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' asterisk1*CLI -- SIP read from 192.168.1.157:5060: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK42700737;rport From: sip:[EMAIL PROTECTED];tag=as558874a4 To: sip:[EMAIL PROTECTED];tag=3a733fa7 Call-ID: [EMAIL PROTECTED] CSeq: 120 REGISTER User-Agent: Grandstream HT488 1.0.2.16 Content-Length: 0 --- The register string ?? Can anyone help me?? Thanks -- Pasqualotto Enrico email: [EMAIL PROTECTED] web: http://www.pasqualotto.org -BEGIN GEEK CODE BLOCK- Version: 3.12 GIT d? s: a-- C+++ UL P L++ E--- W++ N++ o K- w--- O-- M V-- PS+ PE+ Y PGP- t--- 5 X R tv-- b+ DI- D+ G e h++ r+ y+ --END GEEK CODE BLOCK-- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P digium card
What is your setup? There are a lot of variables. How many VOIP trunks do you have? What is your Internet connection? Are you using G.729 for your voip trunks to cut down on bandwidth usage? Anytime you implement a phone system and are using more then just POTS for calls (IE. Voip trunks, remote extensions, etc.), you need to calculate your bandwidth requirements for your Internet connection. Obviously, if you have a slower connection such as xDSL, Cable, T1, you cant have someone on your network file sharing across the Internet and expect good quality VOIP calls. -Dewey From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nora Lavelle Sent: Monday, February 27, 2006 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] TDM400P digium card Thanks dewey. Any feedback on how to debug this issue ? -nora CONFIDENTIALITY NOTICE: This email and any attachments are intended only for the designated recipients. Superior IT Solutions prohibits use, distribution or transmittal by or to an inintended recipient without Superior IT Solution's express written approval. If you are not the intended recipient, please delete this email and notify Superior IT Solutions ([EMAIL PROTECTED])This message (And any attachment) has been scanned by F-Secure and NortonAnti-Virus before leaving our mail server.- ___ --Bandwidth 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] Matching '*'
which doesn't work. So, what exten regex can I use that would catch anything dialled, or how can I stop Asterisk from executing the AGI script a second time when I use _.? I think you can just add an extension h in that context, something like exten = h,1,Hangup hth ___ --Bandwidth 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 501 issues
I am having a couple of (unrelated) problems with my polycom 501. 1. The buddy watch is just not working. It tells me that everyone is online, whether or not they are. Here is an example directory entry for one of the peers (whose phone is not registered). item lnF/ln fnJ/fn ct3062/ct sd1/sd rt3/rt dc/ ad0/ad ar0/ar bw1/bw bb0/bb /item The buddies that are registered do not show on the phone when they are on the phone. In sip.conf (entry for the watching phone--3052) subscribecontext=3058 In extensions.conf: [3058] exten= s,hint,SIP/3058 2.We have a Polycom 501 for testing in our office. Our clients have the other ones. The settings are exactly the same, as all phones use the same sip.cfg file. The sip.conf entries are identical as well (save for username/password and context, of course!) Our clients are calling a few numbers, and even tried our own voicemail system, and no where is key entry heard or recognized. When we dial the same numbers, with the phone in our office, using the same provider, contexts, etc., our key entry is recognized just fine. If the only thing different is the fact that ours is in our office, and the others are located in two different places, what could be the reason that only the phone in our office is working? We don't think it's a DTMF setting, because as we said, the polycoms are all using the same sip.cfg file, and the sip.conf entries are identical. Thanks in advance for your help! ___ --Bandwidth 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] Covad anyone ...
Has anyone done any integration work with Covad's hosted solution ? I am considering Covad's hosted solution and want to be able to use Asterisk to develop some other apps. Anyone else tried this ? how did Covad react. I know they use MGCP. Another thing, the Cisco reseller rep tells me if I have a bunch of 7960's setup for MGCP (for use with Covad) I will need to get these phone hooked up to a Call Manager system to get them to load a SIP image for the first time. That doesn't fit with anything I have been reading from Cisco and other places. Any truth to that ? TIA Alan ___ --Bandwidth 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] AGI Channel Status
I'd like to use the AGI command CHANNEL STATUS to check the status of a channel. However, the dial() command doesn't return -1 until after the call has hung up. If that's the case, how is channel status supposed to return statuses like: status values: 0 Channel is down and available 1 Channel is down, but reserved 2 Channel is off hook 3 Digits (or equivalent) have been dialed 4 Line is ringing 5 Remote end is ringing 6 Line is up 7 Line is busy If dial() doesn't return -1 until after the call is complete, it doesn't seem possible to me that you can check '4 Line is ringing' for example. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI Channel Status
I'd like to use the AGI command CHANNEL STATUS to check the status of a channel. However, the dial() command doesn't return -1 until after the call has hung up. If that's the case, how is channel status supposed to return statuses like: status values: 0 Channel is down and available 1 Channel is down, but reserved 2 Channel is off hook 3 Digits (or equivalent) have been dialed 4 Line is ringing 5 Remote end is ringing 6 Line is up 7 Line is busy If dial() doesn't return -1 until after the call is complete, it doesn't seem possible to me that you can check '4 Line is ringing' for example. Doug Doug, I get the distinct impression that CHANNEL STATUS is to be used independently of the dial() app. At least that's what I see when I read up on AGI, CHANNEL STATUS and dial(). Can you post a snip of the dialplan you've got? I'd like to tinker around with it. Thanks, MC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI Channel Status
MC, But the channel status command is documented as an AGI command itself. If you look at http://www.voip-info.org/wiki-Asterisk+AGI, you'll see the 'channel status' command listed there as an AGI command. I can't post my dial plan, as I don't really have one. Well, I do, and it looks like this: exten = _X.,1,AGI(router.py) Everything is being controlled from the script. The script calls the dial() command. Problem is that dial() doesn't return a -1 (or anything) until after the call is complete. That makes it a BIT tough to check the status of a call. If dial() doesn't return until after the call completes, it means the channel status AGI command is a waste of time. Doug. -Original Message- From: Michael Collins [mailto:[EMAIL PROTECTED] Sent: Monday, February 27, 2006 1:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] AGI Channel Status I'd like to use the AGI command CHANNEL STATUS to check the status of a channel. However, the dial() command doesn't return -1 until after the call has hung up. If that's the case, how is channel status supposed to return statuses like: status values: 0 Channel is down and available 1 Channel is down, but reserved 2 Channel is off hook 3 Digits (or equivalent) have been dialed 4 Line is ringing 5 Remote end is ringing 6 Line is up 7 Line is busy If dial() doesn't return -1 until after the call is complete, it doesn't seem possible to me that you can check '4 Line is ringing' for example. Doug Doug, I get the distinct impression that CHANNEL STATUS is to be used independently of the dial() app. At least that's what I see when I read up on AGI, CHANNEL STATUS and dial(). Can you post a snip of the dialplan you've got? I'd like to tinker around with it. Thanks, MC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] courtesy message calling mobile phones
At 12:07 PM 02/27/2006, you wrote: Can you explain this? What country? In this case it's not asterisk but the telco that has to do the Answer. To every mobile? or just that provider? My knowledge of SS7 is limited, but this has to do with opening the audio path before a call-answered event (which never comes), or even before a call-alerting event. This is also the case where a SIT is generated, and a message like the number you have reached is not in service is played for those not hardcore enough to know the specific error from the sound of the SIT alone. :) -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth 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] Configure DID
I see that you are playing with [EMAIL PROTECTED] How is it going ? Sorry I have not called you. Been very busy. Dovid --- Tele Cost Price Reducer [EMAIL PROTECTED] wrote: Manoj, just look in AMP to Inbound Routing, fill in the DID, define the softphone as extension X and send the call to extension X Mickey On 2/23/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi All, I am a newbie to Asterisk and I was able to install Asterisk and call out. Recently I purchased two DID's, can someone please tell me or point to some links showing how to configure these DID's for SIP based softphones like Express talk? Thanks, Manoj. ___ --Bandwidth 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 __ 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] AGI Channel Status
MC,
I think I worked out that I need to use ${DIALSTATUS} anyway. Don't really see
what 'channel status' is for...
-Original Message-
From: Douglas Garstang
Sent: Monday, February 27, 2006 1:48 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] AGI Channel Status
MC,
But the channel status command is documented as an AGI command itself.
If you look at http://www.voip-info.org/wiki-Asterisk+AGI, you'll see the
'channel status' command listed there as an AGI command.
I can't post my dial plan, as I don't really have one. Well, I do, and it looks
like this:
exten = _X.,1,AGI(router.py)
Everything is being controlled from the script. The script calls the dial()
command. Problem is that dial() doesn't return a -1 (or anything) until after
the call is complete. That makes it a BIT tough to check the status of a call.
If dial() doesn't return until after the call completes, it means the channel
status AGI command is a waste of time.
Doug.
-Original Message-
From: Michael Collins [mailto:[EMAIL PROTECTED]
Sent: Monday, February 27, 2006 1:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] AGI Channel Status
I'd like to use the AGI command CHANNEL STATUS to check the status
of a
channel. However, the dial() command doesn't return -1 until after the
call has hung up. If that's the case, how is channel status supposed
to
return statuses like:
status values:
0 Channel is down and available
1 Channel is down, but reserved
2 Channel is off hook
3 Digits (or equivalent) have been dialed
4 Line is ringing
5 Remote end is ringing
6 Line is up
7 Line is busy
If dial() doesn't return -1 until after the call is complete, it
doesn't
seem possible to me that you can check '4 Line is ringing' for
example.
Doug
Doug,
I get the distinct impression that CHANNEL STATUS is to be used
independently of the dial() app. At least that's what I see when I read
up on AGI, CHANNEL STATUS and dial(). Can you post a snip of the
dialplan you've got? I'd like to tinker around with it.
Thanks,
MC
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[Asterisk-Users] Echo on PRI/BRI?
Howdy: Does echo only occur on analogue PSTN lines, or can it also occur on PRI and BRI lines? If so, for the same reasons? This is a part of our consideration to transition to BRI. Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.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] AGI Scripts Terminate too Soon
Ok, here's a weird one.
I have an AGI script where one user calls another. The call is answered.
Everything is peachy. If the call is terminated by the CALLEE hanging up the
call, then Asterisk returns control back to where the Dial() command left off,
and I can check the return code of Dial(), ${DIALSTATUS} etc. That's all great.
HOWEVER, if the CALLER hangs up the call, it seems as if Asterisk immediately
kills the AGI script. My script seems to terminate immediately and therefore
execution does not continue after the Dial() command. Because of this, I cannot
do any post call processing, or even check the return code from Dial() or
${DIALSTATUS}. I don't know yet, but this may prevent me from being able to put
calls through to voicemail as well.
Anyone seen this? It's a consistent problem.
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Re: [Asterisk-Users] Problem with chan-capi: outgoing calls on two lines
Hello Armin, Am Mo, den 27.02.2006 schrieb Armin Schindler um 20:23: On Mon, 27 Feb 2006, Karsten Wemheuer wrote: Hello, On Mo, 27 Feb 2006, Armin Schindler wrote: This is not a bug, just normal behaviour. chan_capi does not know about the status of the ISDN line, it assumes to be usable when configured. So when you try to dial out chan_capi will choose a channel/line according to internal list of free channels and selects it in the CAPI request. When the driver reports an error via CAPI, chan_capi just signals this error to Asterisk. There is no logic in chan_capi to do something like: If the controller 1 isn't ready, use controller 2. Ok, I didn't know the details of the capi layer. The same happens if the b-channels are already used by another application/device. ... which would not happen on a line in point-to-point mode. No, the line mode doesn't matter. If you have another application running using CAPI, chan_capi also doesn't know about the usage. No, I have no other applications running. And on a point-to-point link there is no other device. As long as there is no outage on the line, I would not have any problems. Thanks for Your information Karsten ___ --Bandwidth 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] Covad anyone ...
Has anyone done any integration work with Covad's hosted solution ? I am considering Covad's hosted solution and want to be able to use Asterisk to develop some other apps. Anyone else tried this ? how did Covad react. I know they use MGCP. Don't know anything about them. Another thing, the Cisco reseller rep tells me if I have a bunch of 7960's setup for MGCP (for use with Covad) I will need to get these phone hooked up to a Call Manager system to get them to load a SIP image for the first time. That doesn't fit with anything I have been reading from Cisco and other places. Any truth to that ? No. One of the last 7960's I bought came with MGCP and I had to jump through hoops to convert it. Its a little different then 7960's with other firmware installed. Since the menues on an mgcp 7960 are totally locked, I had to use a sniffer to detect what IP address the phone was trying to load from, re-IP a linux box to have that IP address, and then follow the instructions on the wiki. It upgraded just fine. If you don't have access to anything that would sniff the packets from the phone (to identify the IP address its trying to load from), it will be very difficult to upgrade it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Channel Status
Douglas Garstang schrieb: If dial() doesn't return until after the call completes, it means the channel status AGI command is a waste of time. Hi, you are right, dial will block, so you won't get the channel status by that method when having an outbound call. You can use the manager. But will have to poll. To avoid polling, I tried to use the manager and parsing the events, but unfortunately the events seems not be reported very reliably in the manager. On high load, some link events imho get lost and are not reported. Roger. ___ --Bandwidth 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] Weird DTMF issue
Ok, this one has me stumped. This setup was working fine Friday and now today it's just stopped working. Details: Dell 2850 running Asterisk 1.2.4. Phones are SIP phones (Cisco 7940s). Timing is done via a WCTDM card (also tried ztdummy.) All traffic in and out of this box to the PSTN is via IAX2. At some point between Friday and today DTMF stopped working right. Specifically, when you call our main # and are at the IVR, only the first digit you dial is recognized. For example if I try to dial 81 this is all I get for debugging: Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: DTMFSubclass: 8 Timestamp: 02123ms SCall: 00020 DCall: 9 [1.2.3.4:4569] Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 02123ms SCall: 9 DCall: 00020 [1.2.3.4:4569] As soon as the 8 is received the Background app stops playing as you'd expect, but it stops recognizing any more digits, and eventually times out and errors out with an invalid extension '8'. Even worse, if you try to dial anybody's direct extensions (2xx) now you end up in the support queue after it times out since the queue is option 2 (yeah I know that's a stupid IVR design, but I had to mimic the old PBX I didn't set up.) I've tried this through two different call paths, one through the PSTN and one direct from my house asterisk system (SIP/IAX2 end-to-end). It behaves the same both ways. The strange part is, while the invalid extension message is being played by Playback() all the digits I hit *are* recognized, as they show up in the iax2 debug output. It's only in the Background() app that this seems to be a problem. Any suggestions would be greatly appreciated. This has our IVR totally busted and I've tried everything I can think of so far. -- Joshua M Thompson [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] Configure DID
Ooops. This was meant to be sent direct and not to the list. Sorry. Dovid --- Dovid Bender [EMAIL PROTECTED] wrote: I see that you are playing with [EMAIL PROTECTED] How is it going ? Sorry I have not called you. Been very busy. Dovid --- Tele Cost Price Reducer [EMAIL PROTECTED] wrote: Manoj, just look in AMP to Inbound Routing, fill in the DID, define the softphone as extension X and send the call to extension X Mickey On 2/23/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi All, I am a newbie to Asterisk and I was able to install Asterisk and call out. Recently I purchased two DID's, can someone please tell me or point to some links showing how to configure these DID's for SIP based softphones like Express talk? Thanks, Manoj. ___ --Bandwidth 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 __ 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 __ 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
[Asterisk-Users] RE: Cisco 7960 upgrade to SIP
I recently upgraded a Cisco 7960 to the SIP firmware, it worked fine without a call-manager. I just put the SIP firmware and associated config files in the TFTP directory of my asterisk server so that the phone could pull the firmware off of my asterisk server via TFTP. It took me about 5 minutes maximum to get the phone working with SIP through asterisk. NO Call-manager was used in the firmware upgrade process. Before SIP my phones where using SCCP (also known as skinny). I imagine that a change from MGCP to SIP should be as painless. KKunzler Message: 16 Date: Mon, 27 Feb 2006 14:14:57 -0600 From: Alan Bunch [EMAIL PROTECTED] Subject: [Asterisk-Users] Covad anyone To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Has anyone done any integration work with Covad's hosted solution ? I am considering Covad's hosted solution and want to be able to use Asterisk to develop some other apps. Anyone else tried this ? how did Covad react. I know they use MGCP. Another thing, the Cisco reseller rep tells me if I have a bunch of 7960's setup for MGCP (for use with Covad) I will need to get these phone hooked up to a Call Manager system to get them to load a SIP image for the first time. That doesn't fit with anything I have been reading from Cisco and other places. Any truth to that ? TIA Alan ___ --Bandwidth 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 Default Ring Volume
The manual mentions that headset, handset, speaker volume are reset between calls to comply with some regulation and there is a setting to prevent this. However it too like the ring volume is completely reset between phone reboots. The file MAC Address-phone.cfg is where the phone would store settings and nothing is stored for ring volume. Polycom could add it in a future firmware version if enough people requested it... --johann Anton Krall wrote: Yep, that much I know but do you know which setting to use? Manual doesn't mention anything. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Wilson Pickett |Sent: Monday, February 27, 2006 10:12 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Polycom Default Ring Volume | |On 2/25/06, Anton Krall [EMAIL PROTECTED] wrote: | Does anybody know how to set polycom's default ring volume ? |Everytime | you restart a polycom phone, ring defaults to a very low volume | setting which is kind of annoying having to set everytime you reboot. | |IIRC, You have to set it in the XML file and reprovision |automatically each time the phone reboots. |___ |--Bandwidth 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] AGI Scripts Terminate too Soon
Douglas Garstang schrieb: ... HOWEVER, if the CALLER hangs up the call, it seems Hi, did you try the dial command option g? I did not neither, but when I understand the voip-wiki right, it might help you. Roger. Voip-wiki page about dial: http://www.voip-info.org/wiki-Asterisk+cmd+Dial ___ --Bandwidth 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] Echo on PRI/BRI?
Does echo only occur on analogue PSTN lines, or can it also occur on PRI and BRI lines? Yes, it can occur on any type of line. If so, for the same reasons? This is a part of our consideration to transition to BRI. It is the result of 4-wire to 2-wire conversion somewhere between your end and the called end. For example, if you originate a call on a PRI/BRI, and you call a telephone number in an analog central office a hybrid conversion will occur somewhere near that terminating CO. That conversion _can_ be a source of echo. In general terms, there are significantly fewer echo issues with PRI's and BRI's then there is with any analog pstn connection. ___ --Bandwidth 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] Covad anyone ...
I have a detailed procedure on migrating from locked MGCP state to SIP, if you get really stuck email me and I will dig it up. Cory Andrews Purchasing Manager ++ VOIPSupply.com A Division of b2 Technologies 454 Sonwil Drive Buffalo, NY 14225 direct - 716.250.3402 mobile - 716.907.4054 email - [EMAIL PROTECTED] AIM - b2Cory - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 27, 2006 4:02 PM Subject: Re: [Asterisk-Users] Covad anyone ... Has anyone done any integration work with Covad's hosted solution ? I am considering Covad's hosted solution and want to be able to use Asterisk to develop some other apps. Anyone else tried this ? how did Covad react. I know they use MGCP. Don't know anything about them. Another thing, the Cisco reseller rep tells me if I have a bunch of 7960's setup for MGCP (for use with Covad) I will need to get these phone hooked up to a Call Manager system to get them to load a SIP image for the first time. That doesn't fit with anything I have been reading from Cisco and other places. Any truth to that ? No. One of the last 7960's I bought came with MGCP and I had to jump through hoops to convert it. Its a little different then 7960's with other firmware installed. Since the menues on an mgcp 7960 are totally locked, I had to use a sniffer to detect what IP address the phone was trying to load from, re-IP a linux box to have that IP address, and then follow the instructions on the wiki. It upgraded just fine. If you don't have access to anything that would sniff the packets from the phone (to identify the IP address its trying to load from), it will be very difficult to upgrade it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco upgrade to SIP was: Covad anyone ...
There is an option that you can add to your dhcp server option 150 IIRC. -Original Message- From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 2/27/06 4:37 PM Has anyone done any integration work with Covad's hosted solution ? I am considering Covad's hosted solution and want to be able to use Asterisk to develop some other apps. Anyone else tried this ? how did Covad react. I know they use MGCP. Don't know anything about them. Another thing, the Cisco reseller rep tells me if I have a bunch of 7960's setup for MGCP (for use with Covad) I will need to get these phone hooked up to a Call Manager system to get them to load a SIP image for the first time. That doesn't fit with anything I have been reading from Cisco and other places. Any truth to that ? No. One of the last 7960's I bought came with MGCP and I had to jump through hoops to convert it. Its a little different then 7960's with other firmware installed. Since the menues on an mgcp 7960 are totally locked, I had to use a sniffer to detect what IP address the phone was trying to load from, re-IP a linux box to have that IP address, and then follow the instructions on the wiki. It upgraded just fine. If you don't have access to anything that would sniff the packets from the phone (to identify the IP address its trying to load from), it will be very difficult to upgrade it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys WIP300 WiFi Phone
Omar A. Sabek wrote: Like BJ, I'm sorry you had bad luck Phil. I have been playing with this phone all weekend, and I have had minor problems. The voice quality is as good as my cisco and polycom sip phones. I asked a friend to guess what kind of phone I was talking on and he said it sounded like a regular home or office phone. I have been very happy with the voice quality. My first day was a huge disappointment. Three crashes, calls wouldn't work over my work's wifi (eventhough it registered ok), short battery time, lost settings after a crash, etc. However, after I went in and cleared my settings back to default, the troubles went away! I'm been using it for over three days without a glitch. So, I would recommend to anybody else who is getting one of these phones, to immediately set all settings back to 'default' (under the Tools menu) before spending too much time configuring it. I reported on the voip-info page dismal talk times but it must have been an anomoly. Today I spoke for over an hour on the phone and still had plenty of juice left. My battery life seems to have improved as well. I don't know if that's was a glitch fixed by setting things back to the defaults, or if cycling the battery is helping. I also have less of a tendency to play with the menus, and the backlight could be a power drainer (it is quite bright). All-in-all this phone is a winner. It works with Asterisk flawlessly. As long as my troubles don't come back, I would agree. I think my phone was shipped to me in a funny state causing it not to work right. It's a winner now. There are some little things I would wish for, but I'm quite happy with it. Phil ___ --Bandwidth 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] Weird DTMF issue
Ok, this one has me stumped. This setup was working fine Friday and now today it's just stopped working. Details: Dell 2850 running Asterisk 1.2.4. Phones are SIP phones (Cisco 7940s). Timing is done via a WCTDM card (also tried ztdummy.) All traffic in and out of this box to the PSTN is via IAX2. At some point between Friday and today DTMF stopped working right. Specifically, when you call our main # and are at the IVR, only the first digit you dial is recognized. For example if I try to dial 81 this is all I get for debugging: Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: DTMFSubclass: 8 Timestamp: 02123ms SCall: 00020 DCall: 9 [1.2.3.4:4569] Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 02123ms SCall: 9 DCall: 00020 [1.2.3.4:4569] As soon as the 8 is received the Background app stops playing as you'd expect, but it stops recognizing any more digits, and eventually times out and errors out with an invalid extension '8'. Even worse, if you try to dial anybody's direct extensions (2xx) now you end up in the support queue after it times out since the queue is option 2 (yeah I know that's a stupid IVR design, but I had to mimic the old PBX I didn't set up.) I've tried this through two different call paths, one through the PSTN and one direct from my house asterisk system (SIP/IAX2 end-to-end). It behaves the same both ways. The strange part is, while the invalid extension message is being played by Playback() all the digits I hit *are* recognized, as they show up in the iax2 debug output. It's only in the Background() app that this seems to be a problem. Any suggestions would be greatly appreciated. This has our IVR totally busted and I've tried everything I can think of so far. It would have been helpfull if you would have posted the few dialplan entries associated with starting the ivr. The following works fine for me for incoming iax2 analog pstn calls: [bus-ivr-main] exten = s,1,Wait,1 exten = s,2,Answer exten = s,3,Set(TIMEOUT(digit)=5) exten = s,4,Set(TIMEOUT(response)=10) exten = s,5,Background(npi-greeting) ; Thanks for calling press 1 for exten = s,6,WaitExten exten = s,7,Goto(bus-ivr-main|s|3) The above is running on /trunk from Feb 20th. If you're funning an older version, the Timeout statements may have a different syntax. You might take a look at 'show application waitexten' and the associated timeout values, etc. ___ --Bandwidth 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] voipstunt can't get call in asterisk
Hi,
does any know why?
i can make call out with my asterisk and voipstunt
but i can't getcall in on my voip in number
i get rejected.
if i use Sipura without asterisk i get in calls
here is my sip.conf
--[general]useragent=nediport=5060context=default;tos=lowdelaydisallow=all
allow=ulaw allow=alaw allow=gsm allow=g726
language=demaxexpiry=50defaultexpiry=30
register = user:[EMAIL PROTECTED]/user
[useruser]type=friendusername=user
secret=passwhost=sip.voipstunt.comfromdomain=sip.voipstunt.comcanreinvite=yesinsecure=verynat=yescontext=incomingsip.voipstunt.comdtmfmode=rfc2833stun=stun.voipstunt.com:3478
[13]type=friendusername=13secret=13callerid="13"
13host=dynamic[EMAIL PROTECTED]dtmfmode=rfc2833canreinvite=yescontext=13
---
my extensions.conf
[general]static=yeswriteprotect=no
[13]include=defaultinclude=outgoinguseruser
exten =13,1,Dial(SIP/13,17,r)exten
=13,2,Answerexten =13,3,Playback(vm-nobodyavail)exten
=13,4,Voicemail(13) exten =13,5,Hangup
[outgoinguseruser]exten = _.,1,Dial(sip/[EMAIL PROTECTED],60)exten
= _.,2,Congestionexten = _.,102,Busy
[incomingsip.voipstunt.com]exten
=user,1,SetCIDName(${CALLERIDNAME})exten =user,2,Dial(Local/[EMAIL PROTECTED])
___
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RE: [Asterisk-Users] AGI Scripts Terminate too Soon
Just tried it. No difference. Here's the console output when the callee hangs up: *CLI -- Executing AGI(SIP/3254102-bb27, ipt/iptrouter.py|FromOnNetPhone) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/ipt/iptrouter.py -- AGI Script Executing Application: (SetMusicOnHold) Options: (default) -- AGI Script Executing Application: (Dial) Options: (SIP/9220402|20|trg) -- Called 9220402 -- SIP/9220402-ca1b is ringing -- SIP/9220402-ca1b answered SIP/3254102-bb27 -- Attempting native bridge of SIP/3254102-bb27 and SIP/9220402-ca1b 1 (ANSWER) -- AGI Script ipt/iptrouter.py completed, returning 0 == Auto fallthrough, channel 'SIP/3254102-bb27' status is 'ANSWER' and here it is when the caller hangs up: *CLI -- Executing AGI(SIP/3254102-0276, ipt/iptrouter.py|FromOnNetPhone) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/ipt/iptrouter.py -- AGI Script Executing Application: (SetMusicOnHold) Options: (default) -- AGI Script Executing Application: (Dial) Options: (SIP/9220402|20|trg) -- Called 9220402 -- SIP/9220402-af98 is ringing -- SIP/9220402-af98 answered SIP/3254102-0276 -- Attempting native bridge of SIP/3254102-0276 and SIP/9220402-af98 == Spawn extension (From_OneEighty, 9220402, 1) exited non-zero on 'SIP/3254102-0276' as you can see, quite different -Original Message- From: Roger Schreiter [mailto:[EMAIL PROTECTED] Sent: Monday, February 27, 2006 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AGI Scripts Terminate too Soon Douglas Garstang schrieb: ... HOWEVER, if the CALLER hangs up the call, it seems Hi, did you try the dial command option g? I did not neither, but when I understand the voip-wiki right, it might help you. Roger. Voip-wiki page about dial: http://www.voip-info.org/wiki-Asterisk+cmd+Dial ___ --Bandwidth 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] AGI Scripts Terminate too Soon
snip HOWEVER, if the CALLER hangs up the call, it seems as if Asterisk immediately kills the AGI script. My script seems to terminate immediately and therefore execution does not continue after the Dial() command. /snip http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI Cheers, Jean-Michel. ___ --Bandwidth 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] AGI Scripts Terminate too Soon
Douglas Garstang schrieb: ... HOWEVER, if the CALLER hangs up the call, it seems Hi, did you try the dial command option g? I did not neither, but when I understand the voip-wiki right, it might help you. Roger. I've used the 'g' option and as far as I can tell it works just the way you want it to - the extension keeps processing even after the destination channel hangs up. I believe the default for the dial() command is to drop the source channel (and rather unceremoniously at that) when the destination channel hangs up. -MC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
