[asterisk-users] Normal ringing tone for the caller, while call waiting.
Hi , When I make a call to an extension, which is on another call, the called party (who is on call waiting) will get a BEEP sound. But the caller gets the normal ringing tone. Is there any way to have a different dialer tone for the Caller, which lets him know that the other person is on a call.. i.e. When A calls B, while B is already on a call with C, Is there a way to let A get a message that B is busy on another call. Thank you. Regards, Najim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to find out one way latency
Hi All, How can I find out One way latency from my PBX to my SIP Trunk Provider. My SIP provider recommends a One way latency of 100ms for good Voice quality. Ping request to their IP Address gives me a response in approx. 260ms. Will that be good enough for a SIP Trunk. Please help. We are trying to sign up with a new SIP Provider. Thanks, Najim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
Thank you Ruben. Is there anything else that I should be concerned about when looking for a SIP provider. ?? Regards, Najim. On Thu, Dec 1, 2011 at 2:34 AM, Ruben Rögels ruben.roeg...@jumping-frog.org wrote: Am 30.11.2011 21:47, schrieb NaJIm: Hi All, How can I find out One way latency from my PBX to my SIP Trunk Provider. My SIP provider recommends a One way latency of 100ms for good Voice quality. Ping request to their IP Address gives me a response in approx. 260ms. Will that be good enough for a SIP Trunk. Please help. We are trying to sign up with a new SIP Provider. Thanks, Najim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Najim, a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. regards, Ruben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
Does that mean I can expect lesser delays with my Voice packets ?? That would be even better. Regards, Najim On Thu, Dec 1, 2011 at 3:42 AM, Steve Edwards asterisk@sedwards.comwrote: Am 30.11.2011 21:47, schrieb NaJIm: Ping request to their IP Address gives me a response in approx. 260ms. Will that be good enough for a SIP Trunk. On Wed, 30 Nov 2011, Ruben Rögels wrote: a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. 'Ping time' is not an accurate predictor of SIP quality. A 'ping' is an ICMP Echo/reply packet and some routers consider them less important than 'data' packets and service them on an 'as resources permit' basis. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
Is there anything else that I should be concerned about, when looking to signup for a SIP provider. ?? Regards, Najim On Thu, Dec 1, 2011 at 4:49 AM, NaJIm getna...@gmail.com wrote: Does that mean I can expect lesser delays with my Voice packets ?? That would be even better. Regards, Najim On Thu, Dec 1, 2011 at 3:42 AM, Steve Edwards asterisk@sedwards.comwrote: Am 30.11.2011 21:47, schrieb NaJIm: Ping request to their IP Address gives me a response in approx. 260ms. Will that be good enough for a SIP Trunk. On Wed, 30 Nov 2011, Ruben Rögels wrote: a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. 'Ping time' is not an accurate predictor of SIP quality. A 'ping' is an ICMP Echo/reply packet and some routers consider them less important than 'data' packets and service them on an 'as resources permit' basis. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
My ping requests show 0% packet loss. How do we find out packet re-ordering.?? Najim. On Thu, Dec 1, 2011 at 5:18 AM, Hans Witvliet aster...@a-domani.nl wrote: On Thu, 2011-12-01 at 04:52 +0530, NaJIm wrote: Is there anything else that I should be concerned about, when looking to signup for a SIP provider. ?? Latency is important, but packet loss also, likewise packet re-ordering. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
WOW.. That is the most complicated Ping I have ever seen.. :) This is the result I got. # ping -f -i .02 -s 180 -Q 0xb8 xx.xx.xx.xx *PING xx.xx.xx.xx (xx.xx.xx.xx) 180(208) bytes of data. . --- xx.xx.xx.xx ping statistics --- 15338 packets transmitted, 15325 received, 0% packet loss, time 352748ms rtt min/avg/max/mdev = 276.499/286.185/310.118/7.248 ms, pipe 15, ipg/ewma 22.999/284.882 ms * The same test with my Present SIP Provider gave me the result below. *10926 packets transmitted, 10913 received, 0% packet loss, time 244048ms rtt min/avg/max/mdev = 289.514/292.668/316.350/2.336 ms, pipe 15, ipg/ewma 22.338/292.941 ms * I suppose the value of mdev is much higher in the first case but 0% packet loss in both the cases. Does this mean that the voice quality is going to be real bad?? Thanks, Najim On Thu, Dec 1, 2011 at 6:33 AM, Adam Moffett adamli...@plexicomm.netwrote: a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. I am probably splitting hairs, but that's not always true because there's no guarantee that the reply traveled the same path as the echo request. If you dig into BGP issues you'll see sometimes that traffic one direction takes a different route than traffic the other direction. I don't know of any simple and accurate way to learn the one way latency so I'm surprised they specified anything other than round trip time. 'Ping time' is not an accurate predictor of SIP quality. A 'ping' is an ICMP Echo/reply packet and some routers consider them less important than 'data' packets and service them on an 'as resources permit' basis. That's possibly maybe true if someone's router or connection is overloaded and they are trying to make up for it with CoS policies while they save up for an upgrade. Otherwise it's an apology for a crappy network. That's the brutally honest truth. You can make a pretty good prediction with ping. sudo ping -f -i .02 -s 180 -Q 0xb8 [ip] gives a tolerable simulation of voip traffic. let it run for awhile, then press ctrl+c and see how many packets were dropped and also check the mdev number. If mdev is low and packet loss is almost nothing then you can expect decent voice quality. It may not be a 100% perfect test, but I'll bet you a vast majority of the time I can do that test and tell you whether it's going to suck. latency by itself with low jitter and no packet loss just means delay. It's a matter of opinion and circumstance how tolerable delay is, but I think your 230ms ping is at the upper edge of what most people can live with. Much more than that and you'll be tempted to say 'over' at the end of sentence. -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterik-users] Installing PRI card
Hi, As Eric mentioned I made my zapata.conf and zaptel.conf to match each other. My *zapata.conf *was *group=1* *switchtype=euroisdn* *signalling=pri_cpe* *callerid=asreceived* *usecallerid=yes* *cidsignalling=dtmf* *cidstart=ring* *context=TEST_EXTERNAL* *channel=1-15* *channel=17-24* This was the error message I was getting when I do *module load chan_zap.so * *[Oct 1 04:29:46] WARNING[17448]: chan_zap.c:905 zt_open: Unable to specify channel 24: Device o r resource busy* *[Oct 1 04:29:46] ERROR[17448]: chan_zap.c:7219 mkintf: Unable to open channel 24: Device or res ource busy* *here = 0, tmp-channel = 24, channel = 24* *[Oct 1 04:29:46] ERROR[17448]: chan_zap.c:10582 build_channels: Unable to register channel '17- 24'* Now I changed *zapata.conf* to * * *group=1* *switchtype=euroisdn* *signalling=pri_cpe* *callerid=asreceived* *usecallerid=yes* *cidsignalling=dtmf* *cidstart=ring* *context=TEST_EXTERNAL* *channel=1-23* Now the error is *[Oct 1 04:33:03] ERROR[17612]: chan_zap.c:9470 start_pri: Unable to open D-channel 24 (Device o r resource busy)* *[Oct 1 04:33:03] ERROR[17612]: chan_zap.c:11382 setup_zap: Unable to start D-channel on span 1* Do I need to mention the Data channel in zapata.conf?? My *zaptel.conf* is below. This was auto generated with *genzaptelconf -svdM * * * *# Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER)* *span=1,1,0,esf,b8zs* *# termtype: te* *bchan=1-23* *dchan=24* I am using a *Digium TE122 Card* . But when I do *dmesg | wcte12xp* I get the following *# dmesg | wcte12xp* * * *wcte12xp :04:00.0: PCI INT A - GSI 21 (level, low) - IRQ 21* *wcte12xp: Setting up global serial parameters for T1* *wcte12xp: Found a Wildcard TE122* *wcte12xp: Span configured for ESF/B8ZS* *wcte12xp: Setting yellow alarm* *wcte12xp0: Missed interrupt. Increasing latency to 4 ms in order to compensate.* *wcte12xp :04:00.0: PCI INT A disabled* *wcte12xp :04:00.0: PCI INT A - GSI 21 (level, low) - IRQ 21* *wcte12xp: Setting up global serial parameters for T1* *wcte12xp: Found a Wildcard TE122* *wcte12xp: Span configured for ESF/B8ZS* *wcte12xp: Setting yellow alarm* *wcte12xp :04:00.0: PCI INT A disabled* *wcte12xp :04:00.0: PCI INT A - GSI 21 (level, low) - IRQ 21* *wcte12xp: Setting up global serial parameters for T1* *wcte12xp: Found a Wildcard TE122* *wcte12xp: Span configured for ESF/B8ZS* *wcte12xp: Setting yellow alarm* *wcte12xp: Span configured for ESF/B8ZS* *wcte12xp: Span configured for ESF/B8ZS* *wcte12xp :04:00.0: PCI INT A disabled* *wcte12xp :04:00.0: PCI INT A - GSI 21 (level, low) - IRQ 21* *wcte12xp: Setting up global serial parameters for T1* *wcte12xp: Found a Wildcard TE122* *wcte12xp: Span configured for ESF/B8ZS* *wcte12xp: Setting yellow alarm* Does the highlighted part mean that the card is setup in T1 mode. ?? ( I do not have physical access to the server. But my support in the remote office says that card is in E1 mode itself.. ) Thanks again for the help. Najim * * On Fri, Sep 30, 2011 at 1:40 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Friday 30 September 2011, NaJIm wrote: Am I getting these error messages due to wrong configurations in my zapata.conf. ?? I have got the following configurations in my zapata-channels.conf. ; Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) B8ZS/ESF RED group=0,11 context=from-pstn switchtype = national signalling = pri_cpe channel = 1-23 group= context=default Your Bearer channels should be 1 to 15 and 17 to 31 (or as high as they go, depending whether or not you paid for a full set of 30 lines), with 16 as the Data channel. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [asterik-users] Installing PRI card
Hi, We have got a new PRI card at one of our Office locations and now I need to install the the device on a remote server. Is there any way to know if the device is loaded already. When I give cat /proc/zaptel/* it returns the following. # cat /proc/zaptel/* Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) B8ZS/ESF RED IRQ misses: 2 1 WCT1/0/1 Clear (In use) RED 2 WCT1/0/2 Clear (In use) RED 3 WCT1/0/3 Clear (In use) RED 4 WCT1/0/4 Clear (In use) RED 5 WCT1/0/5 Clear (In use) RED 6 WCT1/0/6 Clear (In use) RED 7 WCT1/0/7 Clear (In use) RED 8 WCT1/0/8 Clear (In use) RED 9 WCT1/0/9 Clear (In use) RED 10 WCT1/0/10 Clear (In use) RED 11 WCT1/0/11 Clear (In use) RED 12 WCT1/0/12 Clear (In use) RED 13 WCT1/0/13 Clear (In use) RED 14 WCT1/0/14 Clear (In use) RED 15 WCT1/0/15 Clear (In use) RED 16 WCT1/0/16 Clear RED 17 WCT1/0/17 Clear (In use) RED 18 WCT1/0/18 Clear (In use) RED 19 WCT1/0/19 Clear (In use) RED 20 WCT1/0/20 Clear (In use) RED 21 WCT1/0/21 Clear (In use) RED 22 WCT1/0/22 Clear (In use) RED 23 WCT1/0/23 Clear (In use) RED 24 WCT1/0/24 HDLCFCS (In use) RED But when I connect to the console, I am unable to give any ZAP related commands. Does this mean that my device is loaded and I just need to load the module. Or do I need to compile asterisk again?? Any help would be highly appreciated. My asterisk version is Asterisk 1.4.19.2 and I am on a Fedora release 9 server. Thanks, Najim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterik-users] Installing PRI card
Mike, *What version of zaptel are you running? * My Zaptel version is - zaptel-1.4.12.1 *What zaptel commands have you tried? * None of the zaptel commands are working on my CLI. Its like on CLI, none of the commands starting with zap are working. (When I give zap+TAB key nothing shows up) *Have you added any lines to zaptel.conf and zapata.conf? * I am working on a backup server of an already running PBX. In zapata.conf, the only configurations that are there is as below. But I guess those configurations are that of an E1 card and I will have configure it from start. group=1 switchtype=euroisdn signalling=pri_cpe callerid=asreceived usecallerid=yes cidsignalling=dtmf cidstart=ring context=TEST_EXTERNAL channel=1-15 channel=17-31 *Who is your Telco provider and what signalling are they using on the T1? * I am not sure about the signalling they are using. And thanks for the tip on IRQ. As I said I am working on a remote server. I will ask some one over there to change the IRQ value. Regards, Najim On Fri, Sep 30, 2011 at 4:05 AM, Mike Beirne bei...@mgjbnet.com wrote: On 9/29/2011 2:52 PM, NaJIm wrote: IRQ misses: 2 You are risking lots of audio problems if the card shares the IRQ with any other device. Try and go in the BIOS and disable the other device or change the IRQ it is using so that they do not conflict. What version of zaptel are you running? What zaptel commands have you tried? Have you added any lines to zaptel.conf and zapata.conf? Who is your Telco provider and what signalling are they using on the T1? It looks like there is a control channel on 24, but 16 isn't showing the same status as I would expect. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterik-users] Installing PRI card
Hi Eric, This is the error messages I get I try to load the module. *CLI module load chan_zap.so [Sep 30 04:45:57] WARNING[5182]: pbx.c:2979 ast_register_application: Already have an application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, ISDN PRI signalling -- Registered channel 2, ISDN PRI signalling -- Registered channel 3, ISDN PRI signalling -- Registered channel 4, ISDN PRI signalling -- Registered channel 5, ISDN PRI signalling -- Registered channel 6, ISDN PRI signalling -- Registered channel 7, ISDN PRI signalling -- Registered channel 8, ISDN PRI signalling -- Registered channel 9, ISDN PRI signalling -- Registered channel 10, ISDN PRI signalling -- Registered channel 11, ISDN PRI signalling -- Registered channel 12, ISDN PRI signalling -- Registered channel 13, ISDN PRI signalling -- Registered channel 14, ISDN PRI signalling -- Registered channel 15, ISDN PRI signalling -- Registered channel 17, ISDN PRI signalling -- Registered channel 18, ISDN PRI signalling -- Registered channel 19, ISDN PRI signalling -- Registered channel 20, ISDN PRI signalling -- Registered channel 21, ISDN PRI signalling -- Registered channel 22, ISDN PRI signalling -- Registered channel 23, ISDN PRI signalling [Sep 30 04:45:57] WARNING[5182]: chan_zap.c:905 zt_open: Unable to specify channel 24: Device or resource busy [Sep 30 04:45:57] ERROR[5182]: chan_zap.c:7219 mkintf: Unable to open channel 24: Device or resource busy here = 0, tmp-channel = 24, channel = 24 [Sep 30 04:45:57] ERROR[5182]: chan_zap.c:10582 build_channels: Unable to register channel '17-31' Regards, Najim On Fri, Sep 30, 2011 at 4:24 AM, Eric Wieling ewiel...@nyigc.com wrote: Try module load chan_zap.so in the CLI. You should see whatever errors are generated. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of NaJIm Sent: Thursday, September 29, 2011 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [asterik-users] Installing PRI card Hi, We have got a new PRI card at one of our Office locations and now I need to install the the device on a remote server. Is there any way to know if the device is loaded already. When I give cat /proc/zaptel/* it returns the following. # cat /proc/zaptel/* Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) B8ZS/ESF RED IRQ misses: 2 1 WCT1/0/1 Clear (In use) RED 2 WCT1/0/2 Clear (In use) RED 3 WCT1/0/3 Clear (In use) RED 4 WCT1/0/4 Clear (In use) RED 5 WCT1/0/5 Clear (In use) RED 6 WCT1/0/6 Clear (In use) RED 7 WCT1/0/7 Clear (In use) RED 8 WCT1/0/8 Clear (In use) RED 9 WCT1/0/9 Clear (In use) RED 10 WCT1/0/10 Clear (In use) RED 11 WCT1/0/11 Clear (In use) RED 12 WCT1/0/12 Clear (In use) RED 13 WCT1/0/13 Clear (In use) RED 14 WCT1/0/14 Clear (In use) RED 15 WCT1/0/15 Clear (In use) RED 16 WCT1/0/16 Clear RED 17 WCT1/0/17 Clear (In use) RED 18 WCT1/0/18 Clear (In use) RED 19 WCT1/0/19 Clear (In use) RED 20 WCT1/0/20 Clear (In use) RED 21 WCT1/0/21 Clear (In use) RED 22 WCT1/0/22 Clear (In use) RED 23 WCT1/0/23 Clear (In use) RED 24 WCT1/0/24 HDLCFCS (In use) RED But when I connect to the console, I am unable to give any ZAP related commands. Does this mean that my device is loaded and I just need to load the module. Or do I need to compile asterisk again?? Any help would be highly appreciated. My asterisk version is Asterisk 1.4.19.2 and I am on a Fedora release 9 server. Thanks, Najim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterik-users] Installing PRI card
Am I getting these error messages due to wrong configurations in my zapata.conf. ?? I have got the following configurations in my zapata-channels.conf. ; Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) B8ZS/ESF RED group=0,11 context=from-pstn switchtype = national signalling = pri_cpe channel = 1-23 group= context=default Najim On Fri, Sep 30, 2011 at 4:51 AM, NaJIm getna...@gmail.com wrote: Hi Eric, This is the error messages I get I try to load the module. *CLI module load chan_zap.so [Sep 30 04:45:57] WARNING[5182]: pbx.c:2979 ast_register_application: Already have an application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, ISDN PRI signalling -- Registered channel 2, ISDN PRI signalling -- Registered channel 3, ISDN PRI signalling -- Registered channel 4, ISDN PRI signalling -- Registered channel 5, ISDN PRI signalling -- Registered channel 6, ISDN PRI signalling -- Registered channel 7, ISDN PRI signalling -- Registered channel 8, ISDN PRI signalling -- Registered channel 9, ISDN PRI signalling -- Registered channel 10, ISDN PRI signalling -- Registered channel 11, ISDN PRI signalling -- Registered channel 12, ISDN PRI signalling -- Registered channel 13, ISDN PRI signalling -- Registered channel 14, ISDN PRI signalling -- Registered channel 15, ISDN PRI signalling -- Registered channel 17, ISDN PRI signalling -- Registered channel 18, ISDN PRI signalling -- Registered channel 19, ISDN PRI signalling -- Registered channel 20, ISDN PRI signalling -- Registered channel 21, ISDN PRI signalling -- Registered channel 22, ISDN PRI signalling -- Registered channel 23, ISDN PRI signalling [Sep 30 04:45:57] WARNING[5182]: chan_zap.c:905 zt_open: Unable to specify channel 24: Device or resource busy [Sep 30 04:45:57] ERROR[5182]: chan_zap.c:7219 mkintf: Unable to open channel 24: Device or resource busy here = 0, tmp-channel = 24, channel = 24 [Sep 30 04:45:57] ERROR[5182]: chan_zap.c:10582 build_channels: Unable to register channel '17-31' Regards, Najim On Fri, Sep 30, 2011 at 4:24 AM, Eric Wieling ewiel...@nyigc.com wrote: Try module load chan_zap.so in the CLI. You should see whatever errors are generated. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of NaJIm Sent: Thursday, September 29, 2011 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [asterik-users] Installing PRI card Hi, We have got a new PRI card at one of our Office locations and now I need to install the the device on a remote server. Is there any way to know if the device is loaded already. When I give cat /proc/zaptel/* it returns the following. # cat /proc/zaptel/* Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) B8ZS/ESF RED IRQ misses: 2 1 WCT1/0/1 Clear (In use) RED 2 WCT1/0/2 Clear (In use) RED 3 WCT1/0/3 Clear (In use) RED 4 WCT1/0/4 Clear (In use) RED 5 WCT1/0/5 Clear (In use) RED 6 WCT1/0/6 Clear (In use) RED 7 WCT1/0/7 Clear (In use) RED 8 WCT1/0/8 Clear (In use) RED 9 WCT1/0/9 Clear (In use) RED 10 WCT1/0/10 Clear (In use) RED 11 WCT1/0/11 Clear (In use) RED 12 WCT1/0/12 Clear (In use) RED 13 WCT1/0/13 Clear (In use) RED 14 WCT1/0/14 Clear (In use) RED 15 WCT1/0/15 Clear (In use) RED 16 WCT1/0/16 Clear RED 17 WCT1/0/17 Clear (In use) RED 18 WCT1/0/18 Clear (In use) RED 19 WCT1/0/19 Clear (In use) RED 20 WCT1/0/20 Clear (In use) RED 21 WCT1/0/21 Clear (In use) RED 22 WCT1/0/22 Clear (In use) RED 23 WCT1/0/23 Clear (In use) RED 24 WCT1/0/24 HDLCFCS (In use) RED But when I connect to the console, I am unable to give any ZAP related commands. Does this mean that my device is loaded and I just need to load the module. Or do I need to compile asterisk again?? Any help would be highly appreciated. My asterisk version is Asterisk 1.4.19.2 and I am on a Fedora release 9 server
[asterisk-users] Change of default IVR prompt for meetme conference bridge.
Hi, Is it possible to change the default voice prompt for Asterisk meet me conference bridge. We have our own customized recordings for Welcome and PIN request and would like to use that instead of the default Please enter your.. . If I replace the default sound file with my custom file by using the same filename as the default message, will it affect any other applications..?? Thanks, Najim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Confusion with the status of SIP Trunk
Hi, I just wanted to clear a doubt I had. In a SIP trunk, will it show OK status even if only one side of the SIP trunk is configured when we do sip show peers ?? If yes, is there any other way to make sure that the trunk is ready for making calls?? [?] Last day we had a situation here at my Office. There was a fiber cut around our area and one of our leased lines went down. We changed to our secondary line and had to inform our SIP Trunk provider to make the corresponding changes in IP at their end. But even before they did that my trunk status was showing OK and this caused us a lot of confusion. Thank You. [?] Najim 330.png323.png-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confusion with the status of SIP Trunk
Oh.. thank you. That could be the reason. Let me try that. But In fact, I thought qualify = yes is used to send some thing like *keep alive* packets in an already connected trunk to make sure the trunk is still alive. In my case the trunk was completely down, and then it was showing status OK as soon as the Internet came up. Regards, Najim On Thu, Sep 15, 2011 at 2:02 AM, Danny Nicholas da...@debsinc.com wrote: If you use qualify=yes, you should only get OK when the line is functional. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *NaJIm *Sent:* Wednesday, September 14, 2011 3:29 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Confusion with the status of SIP Trunk ** ** Hi, ** ** I just wanted to clear a doubt I had. In a SIP trunk, will it show OK status even if only one side of the SIP trunk is configured when we do sip show peers ?? If yes, is there any other way to make sure that the trunk is ready for making calls?? [image: Description: cid:gtalk.323@goomoji.gmail] ** ** Last day we had a situation here at my Office. There was a fiber cut around our area and one of our leased lines went down. We changed to our secondary line and had to inform our SIP Trunk provider to make the corresponding changes in IP at their end. But even before they did that my trunk status was showing OK and this caused us a lot of confusion. ** ** Thank You. [image: Description: cid:gtalk.330@goomoji.gmail] Najim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users image001.pngimage002.png-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Simultaneous ring on Soft phone and Desk phone.
Hi, In my Office all our users have a Desk phone. Some of the users who are using laptops have a Soft phone too along with their Desk phone. Right now we are using two different extensions for their desk and soft phones. Is it possible to have simultaneous ring for both the extensions (ie. soft phone and desk phone). I tried using Dial(SIP/desk SIP/soft) and it works fine when both the phones are online. But when the soft phone goes offline, none of the phones rings and it says that the user is unavailable. When I tried the above method with both the extensions on the same server, both the phones where ringing simultaneously. But in my case both the extensions are on two different servers and we use SIP trunk to dial between them. We have users located at 7 different Office locations, and each Office has its own PBX for desk phone. All the soft phone extensions are registering to another server. Has anyone setup a similar scenario before.?? Thanks, Najim. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
