Re: [asterisk-users] register = 2345:passw...@sip_proxy/1234
On Fri, 2010-03-19 at 20:14 +0100, Christian Victor wrote: 2010/3/19 tjoen tj...@dds.nl: register = tjoen:mypas...@sip_proxy/1234 [sip_proxy] type=peer host=ekiga.net I guess you need to register to the actual hostname, not the peers name. register = tjoen:mypas...@ekiga.net/1234 Thanks for your response. It is true that actual hostname works, but according to samples peername should work too. Feature broken? -- _ -- 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] Elastix 1.6 continuos ring
Hi Everyone, I have an annoying problem, When get a call from outside to an internal extension, The caller hears continous ring. It should be ring for 5 sec and wait , and goes like this. Iwhere should be the problem? Thanks -- _ -- 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] basic pc to pc voip in lan
Hey everyone The question that i am putting up will sound a bit odd as i am a newbie to asterisk. I have downloaded and install the asterisk and had a look at some of the configuration files like sip.conf, users.conf and extensions.conf. Now my question is that I want to do voip with another pc in LAN, what shall i have to do to implement this and i guess it will be possible without any hardware because at the moment i don't have any. I know that for voip only there are other softwares like skype which are quite easy to operate but I wanna learn implementation of asterisk. Please explain the answer keeping in mind that there is a newbie here. :) Thanks Regards Kartik -- _ -- 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] basic pc to pc voip in lan
Use x-lite softphone. There is no need for any hardware. First you'll setup an extension in sip.conf and then create a dialplan context in extensions.conf so that this extension actually does something. The best way to learn how to do this is by reading chapters 4, 5 and 6 of freely available book Asterisk - The future of telephony. In chapter 4 ignore all about zaptel and start from section Configuring SIP Telephones. -- Zeeshan On 2010-03-20 6:29 AM, kartik manocha koolkarti...@gmail.com wrote: Hey everyone The question that i am putting up will sound a bit odd as i am a newbie to asterisk. I have downloaded and install the asterisk and had a look at some of the configuration files like sip.conf, users.conf and extensions.conf. Now my question is that I want to do voip with another pc in LAN, what shall i have to do to implement this and i guess it will be possible without any hardware because at the moment i don't have any. I know that for voip only there are other softwares like skype which are quite easy to operate but I wanna learn implementation of asterisk. Please explain the answer keeping in mind that there is a newbie here. :) Thanks Regards Kartik -- _ -- 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] 8Port Junghanns BRI card under Dahdi
Hi, I try to get an 8 Port Junghanns BRI card working under dahdi. The card works with zaptel but I have no success under dahdi. I load the module with modprobe wcb4xxp. I dont get any errors but I dont see the spans in /proc/dahdi. The output from dmesg remains empty. I use the following dahdi version: URL: http://svn.digium.com/svn/dahdi/linux/trunk Repository Root: http://svn.digium.com/svn/dahdi Repository UUID: a0bf4364-ded3-4de4-8d8a-66a801d63aff Revision: 8353 Node Kind: directory Schedule: normal Last Changed Author: tzafrir Last Changed Rev: 8347 Last Changed Date: 2010-03-18 13:38:53 +0100 (Thu, 18 Mar 2010) Any idea is welcome. Best regards, Loïc. -- _ -- 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] 8Port Junghanns BRI card under Dahdi
On Sat, 20 Mar 2010, Loic Didelot wrote: Hi, I try to get an 8 Port Junghanns BRI card working under dahdi. The card works with zaptel but I have no success under dahdi. I load the module with modprobe wcb4xxp. I dont get any errors but I dont see the spans in /proc/dahdi. The output from dmesg remains empty. I use the following dahdi version: URL: http://svn.digium.com/svn/dahdi/linux/trunk Repository Root: http://svn.digium.com/svn/dahdi Repository UUID: a0bf4364-ded3-4de4-8d8a-66a801d63aff Revision: 8353 Node Kind: directory Schedule: normal Last Changed Author: tzafrir Last Changed Rev: 8347 Last Changed Date: 2010-03-18 13:38:53 +0100 (Thu, 18 Mar 2010) Any idea is welcome. Did you run dahdi_genconf? Post your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf . Cheers, j -- _ -- 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] 8Port Junghanns BRI card under Dahdi
Hi, I run dahdi_genconf but I think this step can only be successful if the module has been loaded correctly and the spans are visible in /proc/dahdi but this is not the case. Loic. On Sat, 2010-03-20 at 13:21 +, Jeff LaCoursiere wrote: On Sat, 20 Mar 2010, Loic Didelot wrote: Hi, I try to get an 8 Port Junghanns BRI card working under dahdi. The card works with zaptel but I have no success under dahdi. I load the module with modprobe wcb4xxp. I dont get any errors but I dont see the spans in /proc/dahdi. The output from dmesg remains empty. I use the following dahdi version: URL: http://svn.digium.com/svn/dahdi/linux/trunk Repository Root: http://svn.digium.com/svn/dahdi Repository UUID: a0bf4364-ded3-4de4-8d8a-66a801d63aff Revision: 8353 Node Kind: directory Schedule: normal Last Changed Author: tzafrir Last Changed Rev: 8347 Last Changed Date: 2010-03-18 13:38:53 +0100 (Thu, 18 Mar 2010) Any idea is welcome. Did you run dahdi_genconf? Post your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf . Cheers, j -- _ -- 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] Asterisk general Timeout for digits
Hi Guys, I have a need to alter the general timeout in Asterisk. I am wondering if this is something that is hard coded into Asterisk code or if there is a parameter that can be set somewhere. For outbound, I am using x. and hence unless I append a # sign, I would have to wait maybe 5 seconds or so for the call to go through. Is there anywhere in Asterisk that I can change this 5 seconds to let's say 1 second? I understand that there might be the risk of dialing the number unfinished but that's okay with me. Also, for my situation, I can't use specific dial-plans so please guide me to the general timeout parameter if it exists. Thanks, Bruce -- _ -- 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] SIP signal through one IP and media through different IPs
Hi Everyone, I have a provider who is asking me to send SIP signals through 111.111.111.111 and then media through Media 1: 222.222.22.222 and Media 2: 244.244.244.244. This provider authenticates by IP and I think is using Sonus gear and hence they have some load balancer or something... I have always simply done this to work it out: host=111.111.111.111 peer=type and everything worked. But now when I do that I have no audio with call established. I think it's a problem of me not assigning the media IPs. How can I add those to the trunk settings? Thanks, Bruce -- _ -- 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] Asterisk general Timeout for digits
bruce bruce wrote: For outbound, I am using x. and hence unless I append a # sign, I would have to wait maybe 5 seconds or so for the call to go through. Is there You really do need to give us a snippet of the outbound code. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Asterisk general Timeout for digits
As soon as the dialed number matches one of the dial patterns defined in extensions.conf or its included files, asterisk starts dialing it. The wait you have is probably from the trunk provider's side because by default asterisk doesn't start playing the ring tone unless it gets acknoledgement from the provider's side indicating that the call is successfully going through. But even before the above process starts, sip soft phones have their own dialing patterns and timeout values. As soon as your dialed number matches one of them, it is sent to asterisk which does the above. So first you'll have to check your sip phone's dialout pattern and timeout values. -- Zeeshan A Zakaria On 2010-03-20 10:58 AM, Doug Lytle supp...@drdos.info wrote: bruce bruce wrote: For outbound, I am using x. and hence unless I append a # sign, I would ha... You really do need to give us a snippet of the outbound code. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Call Drops while doing assisted transfer from remote location
das sandesh wrote: We are using Yealink T28 phones. Asterisk version: 1.4.21.2, dahdi: 2.2.0.2 This sounds like a few bugs which were opened (and recently closed) related to call transfers. I'm not sure when those bugs were introduced, but upgrading to a newer version of Asterisk may resolve those issues. I suggest you testing 1.4.21.1 on your development platform, replicate your issues, then upgrade to 1.4.30 (again, on your development platform) and see if you can replicate the issues. If you can, then please look at https://issues.asterisk.org to see if you can find an existing open issue. If not, then please file a new issue along with the console output, relevant configurations, and whatever other information would be necessary in order to understand the problem and to reproduce the issue. Thanks! Leif. -- _ -- 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] too much sockets open by asterisk
CHEN XUEQIN wrote: I have a similar problem when using AGI for call control. Also udp port leak for some incomplete call. I wonder if the problem is related to issue 16774. Only way to know would be to reproduce on a development machine, and then try testing the patch on 16774 to see if the issue goes away. Leif. -- _ -- 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 configure caller id
cool dude wrote: hi leif, thx for replying. can u plz ellabroate how to use 'o' optioan in Dial so that callerid should work. There isn't much to elaborate on. Just enable the 'o' option in Dial like any other option, and that should pass through the callerID based on the description you gave. If you're having additional problems with it, then please provide some example dialplan and console output along with a description of what problem you're having. Leif. -- _ -- 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] basic pc to pc voip in lan
Thanks Zeeshan, I'll try that and will revert back to you. On Sat, Mar 20, 2010 at 4:10 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Use x-lite softphone. There is no need for any hardware. First you'll setup an extension in sip.conf and then create a dialplan context in extensions.conf so that this extension actually does something. The best way to learn how to do this is by reading chapters 4, 5 and 6 of freely available book Asterisk - The future of telephony. In chapter 4 ignore all about zaptel and start from section Configuring SIP Telephones. -- Zeeshan On 2010-03-20 6:29 AM, kartik manocha koolkarti...@gmail.com wrote: Hey everyone The question that i am putting up will sound a bit odd as i am a newbie to asterisk. I have downloaded and install the asterisk and had a look at some of the configuration files like sip.conf, users.conf and extensions.conf. Now my question is that I want to do voip with another pc in LAN, what shall i have to do to implement this and i guess it will be possible without any hardware because at the moment i don't have any. I know that for voip only there are other softwares like skype which are quite easy to operate but I wanna learn implementation of asterisk. Please explain the answer keeping in mind that there is a newbie here. :) Thanks Regards Kartik -- _ -- 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 -- _ -- 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] Asterisk general Timeout for digits
Thanks for the input. I am using A2Billing and it takes long time to authenticate PIN number and to dial destination number. If # sign is used then it's a different story and it goes through quick. -Bruce On Sat, Mar 20, 2010 at 11:16 AM, Zeeshan Zakaria zisha...@gmail.comwrote: As soon as the dialed number matches one of the dial patterns defined in extensions.conf or its included files, asterisk starts dialing it. The wait you have is probably from the trunk provider's side because by default asterisk doesn't start playing the ring tone unless it gets acknoledgement from the provider's side indicating that the call is successfully going through. But even before the above process starts, sip soft phones have their own dialing patterns and timeout values. As soon as your dialed number matches one of them, it is sent to asterisk which does the above. So first you'll have to check your sip phone's dialout pattern and timeout values. -- Zeeshan A Zakaria On 2010-03-20 10:58 AM, Doug Lytle supp...@drdos.info wrote: bruce bruce wrote: For outbound, I am using x. and hence unless I append a # sign, I would ha... You really do need to give us a snippet of the outbound code. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 -- _ -- 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] Asterisk general Timeout for digits
Seems like pattern matching needs to be fixed in some config file. Can you give example of a number you dial? -- Zeeshan A Zakaria On 2010-03-20 12:15 PM, bruce bruce bruceb...@gmail.com wrote: Thanks for the input. I am using A2Billing and it takes long time to authenticate PIN number and to dial destination number. If # sign is used then it's a different story and it goes through quick. -Bruce On Sat, Mar 20, 2010 at 11:16 AM, Zeeshan Zakaria zisha...@gmail.com wrote: As soon as the... -- _ -- 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] how to start callerid for india
i belong to india. i am making pbx using sangoma fxo card. i want that when ever call comes to my PSTN line i should see the no from where call is coming. so i have to configures chan_dahdi.conf according to my region. i checked dahdi.conf and in that they have mentioned for india ## ; Hide the name part and leave just the number part of the caller ID ; string. Only applies to PRI channels. ;hidecalleridname=yes ; ; Type of caller ID signalling in use ; bell = bell202 as used in US (default) ; v23 = v23 as used in the UK ; v23_jp = v23 as used in Japan ; dtmf = DTMF as used in Denmark, Sweden and Netherlands ; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi). ; ;cidsignalling=v23 ; ; What signals the start of caller ID ; ring = a ring signals the start (default) ; polarity = polarity reversal signals the start ; polarity_IN = polarity reversal signals the start, for India, ; for dtmf dialtone detection; using DTMF. ; (see doc/India-CID.txt) ; ;cidstart=polarity so i edited chan_dahdi.conf according to my region. ### vi chan_dahdi.conf ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2010-03-18 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;cidstart=ring ;cidstart=polarity ;callerid=asreceived cidsignalling=polarity_IN sendcalleridafter=2 ;Sangoma AU100 [slot:0 bus: span:1] wanpipe1 context=from-zaptel group=0 echocancel=yes signalling = fxs_ks channel = 1 context=from-zaptel group=0 echocancel=yes signalling = fxs_ks channel = 2 now when call comes to PSTN line i am not able to see the no. here is cli log *CLI -- Starting simple switch on 'DAHDI/1-1' [Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 18 (Ring Begin)... [Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 17 (Polarity Reversal)... [Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 17 (Polarity Reversal)... -- Executing [...@from-zaptel:1] Wait(DAHDI/1-1, 2) in new stack -- Executing [...@from-zaptel:2] GotoIfTime(DAHDI/1-1, 23:59-7:59|mon-sun|*|*?closed,s,1) in new stack -- Executing [...@from-zaptel:3] Dial(DAHDI/1-1, SIP/112,60,tT) in new stack == Using SIP RTP CoS mark 5 -- Called 112 -- SIP/112- is ringing == Spawn extension (from-zaptel, s, 3) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' # plz help me out. The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. http://in.yahoo.com/-- _ -- 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 start callerid for india
Does your regular phone shows callerid on this line. If the service provider is sending the callerid, asterisk doesn't have to do anything special to retrieve it. -- Zeeshan On 2010-03-20 1:25 PM, cool dude cool_dudeof...@yahoo.co.in wrote: i belong to india. i am making pbx using sangoma fxo card. i want that when ever call comes to my PSTN line i should see the no from where call is coming. so i have to configures chan_dahdi.conf according to my region. i checked dahdi.conf and in that they have mentioned for india ## ; Hide the name part and leave just the number part of the caller ID ; string. Only applies to PRI channels. ;hidecalleridname=yes ; ; Type of caller ID signalling in use ; bell = bell202 as used in US (default) ; v23 = v23 as used in the UK ; v23_jp = v23 as used in Japan ; dtmf = DTMF as used in Denmark, Sweden and Netherlands ; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi). ; ;cidsignalling=v23 ; ; What signals the start of caller ID ; ring= a ring signals the start (default) ; polarity= polarity reversal signals the start ; polarity_IN = polarity reversal signals the start, for India, ; for dtmf dialtone detection; using DTMF. ; (see doc/India-CID.txt) ; ;cidstart=polarity so i edited chan_dahdi.conf according to my region. ### vi chan_dahdi.conf ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2010-03-18 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;cidstart=ring ;cidstart=polarity ;callerid=asreceived cidsignalling=polarity_IN sendcalleridafter=2 ;Sangoma AU100 [slot:0 bus: span:1] wanpipe1 context=from-zaptel group=0 echocancel=yes signalling = fxs_ks channel = 1 context=from-zaptel group=0 echocancel=yes signalling = fxs_ks channel = 2 now when call comes to PSTN line i am not able to see the no. here is cli log *CLI -- Starting simple switch on 'DAHDI/1-1' [Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 18 (Ring Begin)... [Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 17 (Polarity Reversal)... [Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 17 (Polarity Reversal)... -- Executing [...@from-zaptel:1] Wait(DAHDI/1-1, 2) in new stack -- Executing [...@from-zaptel:2] GotoIfTime(DAHDI/1-1, 23:59-7:59|mon-sun|*|*?closed,s,1) in new stack -- Executing [...@from-zaptel:3] Dial(DAHDI/1-1, SIP/112,60,tT) in new stack == Using SIP RTP CoS mark 5 -- Called 112 -- SIP/112- is ringing == Spawn extension (from-zaptel, s, 3) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' # plz help me out. -- Your Mail works best with the New Yahoo Optimized IE8. Get it NOW!http://in.rd.yahoo.com/tagline_ie8_new/*http://downloads.yahoo.com/in/internetexplorer/ . -- _ -- 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] Asterisk running on a Xen Centos Server challenge!!!
Hi Thanks very much for reply it and helping me out. This is the out put -bash-3.2# ls -l /lib/modules/2.6.18-164.6.1.el5xen/ total 1280 lrwxrwxrwx 1 root root 54 Nov 6 23:31 build - ../../../usr/src/kernels/2.6.18-164.6.1.el5-xen-x86_64 drwxr-xr-x 2 root root 4096 Nov 3 17:31 extra drwxr-xr-x 9 root root 4096 Nov 6 23:31 kernel -rw-r--r-- 1 root root 282675 Nov 6 23:31 modules.alias -rw-r--r-- 1 root root 69 Nov 6 23:31 modules.ccwmap -rw-r--r-- 1 root root 231095 Nov 6 23:31 modules.dep -rw-r--r-- 1 root root147 Nov 6 23:31 modules.ieee1394map -rw-r--r-- 1 root root375 Nov 6 23:31 modules.inputmap -rw-r--r-- 1 root root 12632 Nov 6 23:31 modules.isapnpmap -rw-r--r-- 1 root root 74 Nov 6 23:31 modules.ofmap -rw-r--r-- 1 root root 219500 Nov 6 23:31 modules.pcimap -rw-r--r-- 1 root root 4033 Nov 6 23:31 modules.seriomap -rw-r--r-- 1 root root 132264 Nov 6 23:31 modules.symbols -rw-r--r-- 1 root root 356940 Nov 6 23:31 modules.usbmap lrwxrwxrwx 1 root root 5 Nov 6 23:31 source - build drwxr-xr-x 2 root root 4096 Nov 3 17:31 updates drwxr-xr-x 2 root root 4096 Nov 3 17:31 weak-updates -bash-3.2# This is the other out put. -bash-3.2# ls /usr/src/kernels 2.6.18-164.6.1.el5-xen-i686 2.6.18-164.6.1.el5xen-i686 waiting for you . Thanks very much daniel On 19 Mar 2010, at 8:51 PM, Tzafrir Cohen wrote: On Fri, Mar 19, 2010 at 01:26:43AM +0200, Tzafrir Cohen wrote: On Thu, Mar 18, 2010 at 05:03:12PM -0500, Warren Selby wrote: On Thu, Mar 18, 2010 at 6:56 PM, Daniel Leite de Abreu dlab...@gmail.comwrote: Hi David! Thanks very much for helping me out will all ! Ok i try your tip and @ the moment i still have the same problem but now i have the kernel and the kernel devel the same but wend i try to run make i still get the same erro, do you guys have any idea how to fix it? -bash-3.2# rpm -qa kernel* kernel-xen-devel-2.6.18-164.6.1.el5 kernel-xen-2.6.18-164.6.1.el5 -bash-3.2# After you install the kernel source, you'll need to rerun ./configure. Nope. The dahdi-linux makefile has no ./configure . You may want to run make clean and / or make distclean before rerunning ./configure. Specifically: it will look for: /lib/modules/VERSION/build/.config Where: VERSION is the kernel version string. 2.6.18-164.6.1.el5 in your case. 'build' is a symbolic link to the (often partial) kernel tree. What is the output of: ls -l /lib/modules/2.6.18-164.6.1.el5 What is the output of: ls /usr/src/kernels -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Asterisk running on a Xen Centos Server challenge!!!
Sorry but i did not understand how did you built it? Can you please break up for me? Thanks Dani On 19 Mar 2010, at 12:47 PM, tjoen wrote: On Fri, 2010-03-19 at 01:26 +0200, Tzafrir Cohen wrote: On Thu, Mar 18, 2010 at 05:03:12PM -0500, Warren Selby wrote: On Thu, Mar 18, 2010 at 6:56 PM, Daniel Leite de Abreu dlab...@gmail.comwrote: After you install the kernel source, you'll need to rerun ./configure. Nope. The dahdi-linux makefile has no ./configure . This is how I have built it: DAHDIVERSION=2.2.0.2 build_tools/make_version_h \ include/dahdi/version.h.tmp if cmp -s include/dahdi/version.h.tmp include/dahdi/version.h ; \ then :; \ else \ mv include/dahdi/version.h.tmp include/dahdi/version.h ; \ fi rm -f include/dahdi/version.h.tmp -- _ -- 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] ZTdummy
Hello I have a 4 span PRI board with Zaptel, and Im using it for a long time. In the last days I noticed that the result of zap show status show a ZTDUMMY but I never installed it: o*CLI zap show status Description Alarms IRQ bpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2OK 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 What is that doing ? Also, some people say they cant send me calls through the ZAP trunk... could be because of this ZTDUMMY? Thanks Regards Joao Pereira -- _ -- 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] ZTdummy
You can comment ztdummy out in /etc/sysconfig/zaptel. As for affecting icmo On 2010-03-20 5:16 PM, Joao Gomes Pereira gomespere...@startel.pt wrote: Hello I have a 4 span PRI board with Zaptel, and Im using it for a long time. In the last days I noticed that the result of zap show status show a ZTDUMMY but I never installed it: o*CLI zap show status Description Alarms IRQ bpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2OK 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 What is that doing ? Also, some people say they cant send me calls through the ZAP trunk... could be because of this ZTDUMMY? Thanks Regards Joao Pereira -- _ -- 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] ZTdummy
Ztdummy enabled should not affect incoming calls. On 2010-03-20 5:22 PM, Zeeshan Zakaria zisha...@gmail.com wrote: You can comment ztdummy out in /etc/sysconfig/zaptel. As for affecting icmo On 2010-03-20 5:16 PM, Joao Gomes Pereira gomespere...@startel.pt wrote: Hello I have... -- _ -- 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] 1.6.1.18 - 1.6.2.6 T38 Fax: call drops
Using spandsp-0.0.6-pre17, SendFax on 1.6.1.18 and ReceiveFax on 1.6.2.8. Sip.conf on both sides has t38pt_udptl = yes. -- Executing [...@fax-tx-test:3] SendFAX(SIP/side-sip-0009, /var/spool/asterisk/fax/20091113_1455.tif) in new stack [Mar 20 17:05:34] WARNING[6433]: app_fax.c:178 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. [Mar 20 17:05:34] WARNING[6433]: app_fax.c:772 transmit: Transmission failed -- Executing [...@fax-tx-test:4] Hangup(SIP/side-sip-0009, ) in new stack == Spawn extension (fax-tx-test, s, 4) exited non-zero on 'SIP/side-sip-0009' -- Executing [...@fax-tx-test:1] NoOp(SIP/side-sip-0009, FAXSTATUS: FAILED FAXERROR: The call dropped prematurely FAXMODE: T38) On the receive side: -- Executing [...@incoming-fax:2] ReceiveFAX(SIP/side-sip-0002, /var/spool/asterisk/fax/20100320_1705.tif) in new stack [Mar 20 17:05:34] WARNING[21512]: app_fax.c:223 phase_e_handler: Error transmitting fax. result=48: Disconnected after permitted retries. [Mar 20 17:05:34] WARNING[21512]: app_fax.c:820 transmit: Transmission failed -- Executing [...@incoming-fax:3] Hangup(SIP/side-sip-0002, ) in new stack FAXSTATUS: FAILED FAXERROR: Disconnected after permitted retries FAXMODE: T38) Sip audio calls work fine over side-sip. Any suggestions welcome. sean -- _ -- 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 start callerid for india
Hi, Edit your logger.conf, set messages in debug mode, make test incoming and outgoing calls. Copy the log in message dirz3* and post. Goke On 3/20/10, Zeeshan Zakaria zisha...@gmail.com wrote: Does your regular phone shows callerid on this line. If the service provider is sending the callerid, asterisk doesn't have to do anything special to retrieve it. -- Zeeshan On 2010-03-20 1:25 PM, cool dude cool_dudeof...@yahoo.co.in wrote: i belong to india. i am making pbx using sangoma fxo card. i want that when ever call comes to my PSTN line i should see the no from where call is coming. so i have to configures chan_dahdi.conf according to my region. i checked dahdi.conf and in that they have mentioned for india ## ; Hide the name part and leave just the number part of the caller ID ; string. Only applies to PRI channels. ;hidecalleridname=yes ; ; Type of caller ID signalling in use ; bell = bell202 as used in US (default) ; v23 = v23 as used in the UK ; v23_jp = v23 as used in Japan ; dtmf = DTMF as used in Denmark, Sweden and Netherlands ; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi). ; ;cidsignalling=v23 ; ; What signals the start of caller ID ; ring= a ring signals the start (default) ; polarity= polarity reversal signals the start ; polarity_IN = polarity reversal signals the start, for India, ; for dtmf dialtone detection; using DTMF. ; (see doc/India-CID.txt) ; ;cidstart=polarity so i edited chan_dahdi.conf according to my region. ### vi chan_dahdi.conf ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2010-03-18 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;cidstart=ring ;cidstart=polarity ;callerid=asreceived cidsignalling=polarity_IN sendcalleridafter=2 ;Sangoma AU100 [slot:0 bus: span:1] wanpipe1 context=from-zaptel group=0 echocancel=yes signalling = fxs_ks channel = 1 context=from-zaptel group=0 echocancel=yes signalling = fxs_ks channel = 2 now when call comes to PSTN line i am not able to see the no. here is cli log *CLI -- Starting simple switch on 'DAHDI/1-1' [Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 18 (Ring Begin)... [Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 17 (Polarity Reversal)... [Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 17 (Polarity Reversal)... -- Executing [...@from-zaptel:1] Wait(DAHDI/1-1, 2) in new stack -- Executing [...@from-zaptel:2] GotoIfTime(DAHDI/1-1, 23:59-7:59|mon-sun|*|*?closed,s,1) in new stack -- Executing [...@from-zaptel:3] Dial(DAHDI/1-1, SIP/112,60,tT) in new stack == Using SIP RTP CoS mark 5 -- Called 112 -- SIP/112- is ringing == Spawn extension (from-zaptel, s, 3) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' # plz help me out. -- Your Mail works best with the New Yahoo Optimized IE8. Get it NOW!http://in.rd.yahoo.com/tagline_ie8_new/*http://downloads.yahoo.com/in/internetexplorer/ . -- _ -- 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 -- Sent from my mobile device -- _ -- 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] Voicemail, Asterisk and Grandstream BT200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm testing with a Grandstream BT200 telephone and, according to I read, it has a LED that blinks if for that extension messages were left. In Voice Mail UserID, under the ACCOUNT tab, I put *100 that is the extension in which my Asterisk answer the voicemail service and if then I press MESSAGE button, the telephone communicates with Asterisk and, after to introduce the password, it indicates to me that I have messages. But the luminous indicator does not work. It is necessary to configure something special for this? It can be that it doesn't work because there is to introduce one password previously? Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkulXlgACgkQZpa/GxTmHTfQBACfUkqoST6HRgpsXwcBZpXfLdan UaoAn2peX4pmoe3GlgoBL9GcOBxmg9UR =0RBM -END PGP SIGNATURE- -- _ -- 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] Early audio problem in chan_dahdi
Hello, if have a problem since I switched to asterisk-1.6: When making an outgoing call through chan_dahdi, I cannot hear anymore early audio, the asterisk generated sound (as defined in indications.conf) is played. Thus, I cannot hear announcements by the operator, and when the line is busy, sometimes I can hear first the ringing indication by asterisk, and some moments later the busy. I already tried both in chan_dahdi.conf: callprogress = yes and callprogress = no No difference. What I'm doing wrong? Roger. -- _ -- 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
