Re: [asterisk-users] Getting 'username' of sip peer
Yes, setting a fullname=xxx in peer definition sets the CALLERID(name) But if the peer sets it own callerid then will it override this value? On Wed, May 26, 2010 at 10:52 PM, Danny Nicholas da...@debsinc.com wrote: I might be wrong, but I think that adding fullname=xxx to the context will populate CALLERID(name) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D Sent: Wednesday, May 26, 2010 12:18 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting 'username' of sip peer Hello, I have a few entries for sip peers in sip.conf with different name and username, like [TestSIPUser] type=peer host=dynamic username=testuser secret=1234 context=test_context [TestNewUser] type=peer host=dynamic username=newsipuser secret=3456 context=test_context When a call is made from any of these peers I want to get the username of the peer. for eg:- If a call is being made from 'TestSIPUser' then I want to be able to get the value 'testuser' Is it possible to get the value of 'username' of the peer in the dialplan using some application/function ? Thanks, Deepesh -- _ -- 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] Getting 'username' of sip peer
Thanks. Got this working by using setvar=variable=value in the peer
definition
SIPCHANINFO(peername) is giving me the 'name' of the peer i.e.
'TestSIPUser' and not the 'username'.
On Wed, May 26, 2010 at 11:20 PM, Jared Smith jsm...@digium.com wrote:
On Wed, 2010-05-26 at 22:48 +0530, Deepesh D wrote:
When a call is made from any of these peers I want to get the username
of the peer.
for eg:- If a call is being made from 'TestSIPUser' then I want to be
able to get the value 'testuser'
I can think of two ways of doing this. The first is to use the
SIPCHANINFO() dialplan function, like this:
exten=123,1,Verbose(0,The call came from ${SIPCHANINFO(peername)})
The other option is to use the setvar=variable=value setting in the
peer definition in sip.conf. For example, if you add
setvar=USERID=jsmith in a user/peer/friend definition, Asterisk would
automagically create a channel variable named USERID with a value of
jsmith every time this device made a call into Asterisk.
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Digium, Inc.
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Re: [asterisk-users] Jack in /usr/local/ means failure for asterisk
Hi Motiejus! I'll look for JACK's configure script and send it off-list, unless someone else here wants it? Now about my programs. Scenario: Start Asterisk. Then directly use CLI to dial. And if possible use asterisk only to pick up calls. problem: I didn't find an easy way to let the phone ring with asterisk only. I either could directly answer or put the incoming call elsewhere. So I wrote a simple application to be used via system() in a diaplan. It listens on a simple UDP - telnet - port for a connection. When that connection comes, the phone will be answered. A status will be returned, that you can query in the dialplan. If you don't connect, the program will stop after a while and return a different status, so the call will be sent to voice mail. In a parallel thread the program executes an audioplayer with a configureable audiofile, to let it reall ring. It would be stupid if you couldn't hear anything ringing. Even better; Because the audiofile is a parameter, you don't even have to look at a caller display - if someone froma preconfigured group calls -, because you can set a specific ringtone for them. The other mini-script I wrote, contains a simple telnet line. But because that is too long to type, when answering, the script is simply called dk. :-) Quite special this application, I grant you that, but handy if you like asterisk on its own. That way, if asterisk works with jack, I can have a fully fledged pbx with a commandline interface. :-) Now off to JACK and its configure script... Best wishes Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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] Meetmee user introduction disabled
I updated Asterisk to 1.6.2.7 and now the user introduction in the meetme application is no longer working: [May 27 09:26:51] WARNING[2407]: channel.c:4034 ast_request: No channel type registered for 'DAHDI' -- Created MeetMe conference 1023 for conference '800' [May 27 09:26:51] WARNING[2407]: app_meetme.c:3640 find_conf: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) [May 27 09:26:51] WARNING[2407]: app_meetme.c:3646 find_conf: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) The conference itself seems to work. The box does not have any special hardware so dadhi is only needed for timing and as I understood for conference mixing. As part of the update I also updated the dadhi modules and the kernel (all with yum update): kmod-dahdi-linux-2.3.0.1-1_centos5.2.6.18_194.3.1.el5 kmod-dahdi-linux-2.2.1-1_centos5.2.6.18_164.11.1.el5 First I noted that dahdi_dummy is no longer present in kmod-dahdi-linux-2.3.0.1-1. Reverting back to kmod-dahdi-linux-2.2.1-1 solved that issue, now the module is loaded again. lsdahdi ### Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: Linux26) 1 (MASTER) lsmod|grep dahdi_dummy dahdi_dummy 8612 0 dahdi 194504 14 dahdi_dummy,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp Asterisk however shows the same warning message. So I guess something has changed in the meetme application itself and it does not seem to use dahdi_dummy anymore? What steps can I take? Theo -- _ -- 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] Adding a context from the console
Should I log this as a bug since it doesn't work? Regards Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer Sent: 20 May 2010 16:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Adding a context from the console Hi, this didn't seem to work. Is there something I am missing? dialplan add extension 1234,1,NoOp,hello into default Extension '1234,1,NoOp,hello' added into 'default' context -- Added extension '1234' priority 1 to default (0x8e8f520) dialplan add extension 1234,1,NoOp,hello into test Failed to add '1234,1,NoOp,hello' extension into 'test' context Regards Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer Sent: 19 May 2010 16:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Adding a context from the console Many thanks. Regards Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: 19 May 2010 16:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Adding a context from the console On Wednesday 19 May 2010 02:28:02 Lee Archer wrote: Hi, is it possible to add a context from the console using the dialplan command? Yes, just add an extension to it. The context will be created as needed. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Wed, May 26, 2010 at 09:52:52PM +0200, Vincent wrote:
On Wed, 26 May 2010 17:30:08 +0200, Vincent codecompl...@free.fr
wrote:
More information, as I investigate:
For those having the same issue, here's what I learned:
1. In /etc/modprobe.d/dahdi.blacklist.conf, blacklist the netjet
driver:
blacklist netjet
This is a bug of the netjet module. It should not try to handle those
devices. While they use the netjet chipset, they are not the ISDN BRI
devices drivven by it.
Looking at drivers/isdn/hardware/mISDN/netjet.c:
/* We cannot select cards with PCI_SUB... IDs, since here are cards with
* SUB IDs set to PCI_ANY_ID, so we need to match all and reject
* known other cards which not work with this driver - see probe
* function */
static struct pci_device_id nj_pci_ids[] __devinitdata = {
{ PCI_VENDOR_ID_TIGERJET, PCI_DEVICE_ID_TIGERJET_300,
PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0},
{ }
};
MODULE_DEVICE_TABLE(pci, nj_pci_ids);
And indeed, nj_probe() above has:
struct tiger_hw *card;
if (pdev-subsystem_vendor == 0x8086
pdev-subsystem_device == 0x0003) {
pr_notice(Netjet: Digium X100P/X101P not handled\n);
return -ENODEV;
}
if (pdev-subsystem_vendor == 0x55
pdev-subsystem_device == 0x02) {
pr_notice(Netjet: Enter!Now not handled yet\n);
return -ENODEV;
}
But sadly, only those.
If nobody beats me to it, I'll try submitting a (untested) extended list
of exceptions over the weekend.
For an initial list:
grep -i e159 xpp/perl_modules/Dahdi/Hardware/PCI.pm
from the top-level directory of DAHDI-tools.
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Re: [asterisk-users] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
DON'T RUN dahdi_genconf, as it overwrites system.conf. Yes. dahdi_genconf reads /etc/dahdi/genconf_parameters and writes /etc/dahdi/system.conf and /etc/asterisk/dahdi_channels.conf. You can set the country as lc_country fr in /etc/dahdi/genconf_parameters. 1. When I run dahdi_genconf: /usr/sbin/dahdi_genconf: Failed to open /etc/asterisk/dahdi-channels.conf: No such file or directory Do you have asterisk installed? You neet at least to mkdir /etc/asterisk. Ciao Leo -- _ -- 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] N900 video with Asterisk?
Anyone using an N900 with asterisk yet? Had mine for a while now and VoIP (voice) has been working really well, but the new firmware update brings SIP Video calling too - so just given it a go... The fly in the ointment is that I only have a Grandstream GXV3000 to test it with ... And it worked - sort of. If I call the N900 from the Grandstream it's OK (H263 and H264 seems supported), and I get video both ways, but if I call the GXV3000 from the N900, then I don't get any video - sound is OK. Even after pushing the 'video' and camera buttons on the N900 I don't get anything. So sort of scratching my head here - I have had these sorts of issues trying to get Ekiga going with the GXV - was a codec issue with Ekiga until I got the extra codecs, but not sure what's going on here. Actually, I'll try it with Ekiga in a bit... But if anyone else has had a play, let me know how you got on! Cheers, Gordon -- _ -- 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] BRI card(B800P) doesn's work with DAHDI(wcb4xxp) in NT mode
On Thu, May 27, 2010 at 11:12:05AM +0800, Michael wrote: Dear Supports, I was attempting to install BRI Card(OpenVox B800P) with wcb4xxp in NT mode .But I can not make it worked! Could you please give me some hints? Thanks in advance! What version of Asterisk? BRI NT PtMP is not supported in current released versions. Try using PtP instead, if applicable (or Asterisk trunk, if you really want to help us test it ;-) -- 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
Re: [asterisk-users] routing of calls
On Wed, May 26, 2010 at 04:41:57PM +0100, salaheddine elharit wrote:
Hello All
i have set all extensions for 2 providers in dialplan.conf and
extensions.conf
What's dialplan.conf ?
the problem is all numbers take the same provider
when i change the g1 with g2 all the phones numbers take the secend
provider
; Outbound dial context
[aheeva_ccs]
; If we are dialing out through another Asterisk, sometimes when a call is
not
; answered the DIALSTATUS gets set to CANCEL and Asterisk just aborts the
DIAL
; and jumps directly to the h extension without continuing processing in the
; dialplan after the Dial application, which means that we do not send the
; DIALSTATUS to the CCS server after the dial. This is why we need to
capture
; here in the h extension and send a NOANSWER.
exten = h,1,NoOp(ds= ${DIALSTATUS});
exten = h,2,GotoIf($[${DIALSTATUS} = ANSWER]?6:3)
exten = h,3,GotoIf($[${DIALSTATUS} = CANCEL]?4:5)
exten =
h,4,AHEventsProxy(NOANSWER:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH})
exten =
h,5,AHEventsProxy(MSG_TYPE_TERMINATE_CALL:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH}:${AH_AGENTID})
What is AHEventsProxy()? Is that a dialplan application? From what
module does it come?
exten = h,6,Hangup
exten = _OUT.,1,NoOp(AHEEVA1 Variables:
AH_PHONE_NUMBER=[${AH_PHONE_NUMBER}] AH_QUEUE=[${AH_QUEUE}]
AH_URL=[${AH_URL}] AH_RECORDID=[${AH_RECORDID}]
AH_AMD_REQUIRED=[${AH_AMD_REQUIRED}] AH_CALLERID=[${AH_CALLERID}]
AHEEVA_TRACKNUM=[${AHEEVA_TRACKNUM}] AH_LEAVE_MESSAGE=[${AH_LEAVE_MESSAGE}])
exten = _OUT.,2,SetCallerId(${AH_CALLERID})
exten = _OUT.,3,Dial(Zap/g1/${AH_PHONE_NUMBER},30)
exten = _OUT.,4,NoOp(Dial Status=[${DIALSTATUS}] Hangup
Cause=[${HANGUPCAUSE}])
exten = _OUT.,5,GotoIf($[${DIALSTATUS} = CHANUNAVAIL ${HANGUPCAUSE}
= 16]?6:8)
exten =
_OUT.,6,AHEventsProxy(MSG_TYPE_CALL_SIT:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH})
exten = _OUT.,7,Goto(9)
exten =
_OUT.,8,AHEventsProxy(${DIALSTATUS}:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH})
exten = _OUT.,9,NoOp()
thanks a lot
2010/5/26 Doug Lytle supp...@drdos.info
salaheddine elharit wrote:
G2 is for the second provider and g1 for the first provider even I
configured the extensios.conf I have some calls passed from g1
instead g2
Any help please will be appreciated
Maybe if you asked a question, something could help. But, as it is
stated now, I'm have no idea as to what you want help with.
Doug
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Re: [asterisk-users] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Thu, 27 May 2010 12:29:05 +0300, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: This is a bug of the netjet module. It should not try to handle those devices. While they use the netjet chipset, they are not the ISDN BRI devices drivven by it. Thanks for the explanation. On this exact same hardware, I didn't have this problem with Dahdi/Zaptel 1.4. -- _ -- 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] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Thu, 27 May 2010 11:41:09 +0200, Leonardo Pistone l.pist...@sispac.it wrote: Yes. dahdi_genconf reads /etc/dahdi/genconf_parameters and writes /etc/dahdi/system.conf and /etc/asterisk/dahdi_channels.conf. Thanks for the tip. Do you have asterisk installed? You neet at least to mkdir /etc/asterisk. Nope, and running mkdir /etc/asterisk solved this issue. There's one thing left: # /etc/init.d/dahdi restart Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wctdm: [ OK ] /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting Running dahdi_cfg: [ OK ] I assume this reference to astribank is due to default settings. How can I remove unneeded drivers/modules? Thank you. -- _ -- 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] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Thu, May 27, 2010 at 03:03:21PM +0200, Vincent wrote: On Thu, 27 May 2010 12:29:05 +0300, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: This is a bug of the netjet module. It should not try to handle those devices. While they use the netjet chipset, they are not the ISDN BRI devices drivven by it. Thanks for the explanation. On this exact same hardware, I didn't have this problem with Dahdi/Zaptel 1.4. Older kernel did not have the netjet module? -- 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
Re: [asterisk-users] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Thu, 27 May 2010 16:12:57 +0300, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Thanks for the explanation. On this exact same hardware, I didn't have this problem with Dahdi/Zaptel 1.4. Older kernel did not have the netjet module? Yup, that could be the reason. Anyway, problem solved :-) Thank you. -- _ -- 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] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Thu, 27 May 2010 15:09:45 +0200, Vincent codecompl...@free.fr wrote: /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting Running dahdi_cfg: [ OK ] it's harmless. but it's a symtom of building dahdi-tools without libusb https://issues.asterisk.org/view.php?id=17189 -- _ -- 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] Call Timeout on 302 Redirect
Currently running Asterisk 1.6.2.6 with Polycom 550 phones with the latest SIP firmware. We use the forward functionality on the phones (primarily forward on no answer), and it works very well with one caveat: call timeout. When the phone redirects (usually after 2 rings), asterisk still rolls the call to voicemail after whatever is the set timeout. So, if I have it set to something like 20 seconds, there isn't enough time for the second phone (usually a mobile phone) to ring, and if I set it longer then the users who don't use call forwarding complain that it takes too long for their voicemail to pickup. I have been searching around for information/fixes for this problem, but I am either not using the right terms, or there just isn't anything out there. I did stumble across this: https://issues.asterisk.org/view.php?id=17340 which makes me think that the functionality simply doesn't exist in asterisk to have it both ways. What I mean by that is that I would like something around a 20 second timeout, but if they forward (when asterisk gets the 302 message back), I would like that timeout to reset and give another 20 seconds. Is this possibly, is there another way people are handling this, am I just chasing something not possible? Thanks. -Jay -- _ -- 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] Dahdi problems with kernel 2.6.32
On 05/26/2010 08:00 PM, cov...@ccs.covici.com wrote: From another thread, I blacklisted netjet and now things are working. But I wonder what is going on here and where did netjet come from -- it doesn't look like an dahdi module to me. It comes from mISDN. It is a very badly misbehaving module. IIRC, it wildcards a large portion of tigerjet PCI IDs. -- _ -- 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] Meetmee user introduction disabled
On Thu, May 27, 2010 at 4:05 AM, Theo Band theo.b...@greenpeak.com wrote: First I noted that dahdi_dummy is no longer present in kmod-dahdi-linux-2.3.0.1-1. Not exactly true. myhost01 asterisk # lsmod | grep dahdi dahdi_dummy 5812 0 dahdi_transcode 8968 1 wctc4xxp dahdi_voicebus 42048 2 wctdm24xxp,wcte12xp dahdi 198992 24 dahdi_dummy,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 4096 2 wctdm24xxp,dahdi myhost01 asterisk # dmesg | grep dahdi dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.3.0 Reverting back to kmod-dahdi-linux-2.2.1-1 solved that issue, now the module is loaded again. I suppose it would. I got dahdi_dummy with 2.3.0 by analyzing how the build process worked, and doing some tricks. I could see dahdi_dummy.c was in the package but it wasn't getting built. Here's the trick. If you pull down the combined dahdi package, extract it, cd into the extracted top-level folder cd linux (which is the dahdi proper stuff) make MODULES_EXTRA=dahdi_dummy That worked for me. Do the make install too. asktest01 linux # make MODULES_EXTRA=dahdi_dummy make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi/firmware' make -C /lib/modules/2.6.28.9/build SUBDIRS=/usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi DAHDI_INCLUDE=/usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/include DAHDI_MODULES_EXTRA=dahdi_dummy.o HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: Entering directory `/usr/src/linux-2.6.28.9' CC [M] /usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi/dahdi_dummy.o Building modules, stage 2. MODPOST 31 modules CC /usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi/dahdi_dummy.mod.o LD [M] /usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi/dahdi_dummy.ko make[1]: Leaving directory `/usr/src/linux-2.6.28.9' It seems that these days you need to provide extra arguments to get dahdi_dummy, and it's getting filtered by default. -- _ -- 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] IAX2 Call Transfer
Hi all, i do have the following setup Incoming call over DAHDI - to another machine using IAX2 - Agent at this machine starts an attended transfer to an external number This new initiated call does go over IAX2 - the machine the original call came in - DAHDI out into the world. Agent does release the channel - so asterisk does bridge the channel which is at the agent machine with the channel the agent created for the new outbound call. So as far as i do understand this right - the media path after the transfer is still DAHDI - machine 1 - machine agent - machine 1 - DAHDI. Or is the IAX2 Protocol smart enough to detect this - and does not send the media across the line ? If IAX2 is not smart enough - how could i make this possible ? I already thought that something like this could do that trick: Incoming DAHDI channel - do dial IAX2 trunk on the same machine - so it is a native IAX2 channel - go to agent machine - agent does create outbound IAX2 channel - agent does transfer - now asterisk can handle it native - and will bridge channels on the first machine. Would this work ? best regards, Wolfgang Pichler -- _ -- 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] GoogleTalk to Asterisk - choosing voice menu options
GoogleTalk connects ok to Asterisk 1.6.2.7 but how can you choose voice menu options (press 1 for Bob, press 2 for Betty, ...) from the GT client? (There is no dial pad in the Windows GT client, but what you type in the message box does show up on the console as an incoming Jabber message.) Is there a way? Thanks all! -- _ -- 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] Adding a context from the console
On Thursday 27 May 2010 03:55:05 Lee Archer wrote:
On Wednesday 19 May 2010 16:44 Tilghman Lesher wrote:
On Wednesday 19 May 2010 02:28:02 Lee Archer wrote:
Hi, is it possible to add a context from the console using the
dialplan command?
Yes, just add an extension to it. The context will be created as
needed.
Hi, this didn't seem to work. Is there something I am missing?
dialplan add extension 1234,1,NoOp,hello into default
Extension '1234,1,NoOp,hello' added into 'default' context
-- Added extension '1234' priority 1 to default (0x8e8f520)
dialplan add extension 1234,1,NoOp,hello into test
Failed to add '1234,1,NoOp,hello' extension into 'test' context
File a bug on this. The patch is as simple as (attached).
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
Index: pbx/pbx_config.c
===
--- pbx/pbx_config.c(revision 266141)
+++ pbx/pbx_config.c(working copy)
@@ -1543,6 +1543,7 @@
if (!app_data)
app_data=;
+ ast_context_find_or_create(NULL, argv[5], registrar);
if (ast_add_extension(argv[5], argc == 7 ? 1 : 0, exten, iprior, NULL,
cidmatch, app,
(void *)strdup(app_data), ast_free_ptr, registrar)) {
switch (errno) {
--
_
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Re: [asterisk-users] Adding a context from the console
I assume this patch is for 1.6X since I find no code similar to this in 1.4.30? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Thursday, May 27, 2010 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Adding a context from the console On Thursday 27 May 2010 03:55:05 Lee Archer wrote: On Wednesday 19 May 2010 16:44 Tilghman Lesher wrote: On Wednesday 19 May 2010 02:28:02 Lee Archer wrote: Hi, is it possible to add a context from the console using the dialplan command? Yes, just add an extension to it. The context will be created as needed. Hi, this didn't seem to work. Is there something I am missing? dialplan add extension 1234,1,NoOp,hello into default Extension '1234,1,NoOp,hello' added into 'default' context -- Added extension '1234' priority 1 to default (0x8e8f520) dialplan add extension 1234,1,NoOp,hello into test Failed to add '1234,1,NoOp,hello' extension into 'test' context File a bug on this. The patch is as simple as (attached). -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] OpenVox B200P and D410P under Asterisk 1.6
Hello all- My client has purchased these two OpenVox cards and I'm configuring a system with Asterisk 1.6. In the past I have used bristuff and libpri with older versions of Asterisk, but now I would like to upgrade to Asterisk 1.6. Question, should I be using mISDN or libpri for these cards when they are in the same system, or does DAHDI now support both cards under asterisk 1.6 reliably? I'm especially concerned about the OpenVox B200P as I haven't used it before. 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] How to have Asterisk respond from the IP address used for registration
Hi, I have a test server with 2 NICs, each with it own IP address. Let`s say 192.168.1.2 and 192.168.1.3. I would like some phones to register by using 192.168.1.2 and some by using 192.168.1.3 as the address. Since the default IP is 192.168.1.2, that is the only working address. Every phone connecting to 192.168.1.3 fails to register, presumably because Asterisk answers back from 192.168.1.2 and the phone doesn't recognize this as the correct SIP server. I am using 1.4.31. Is there any way to have Asterisk answer from the IP address used instead of using the default one? Regards, 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
Re: [asterisk-users] How to have Asterisk respond from the IP addressused for registration
2 things to try - (1) set bindaddr in sip.conf to 0.0.0.0 instead of 192.168.1.2 - in theory this will let * use both cards (2) start second instance of asterisk bound to 192.168.1.3 - probably the approach with the better chance of success. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, May 27, 2010 2:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] How to have Asterisk respond from the IP addressused for registration Hi, I have a test server with 2 NICs, each with it own IP address. Let`s say 192.168.1.2 and 192.168.1.3. I would like some phones to register by using 192.168.1.2 and some by using 192.168.1.3 as the address. Since the default IP is 192.168.1.2, that is the only working address. Every phone connecting to 192.168.1.3 fails to register, presumably because Asterisk answers back from 192.168.1.2 and the phone doesn't recognize this as the correct SIP server. I am using 1.4.31. Is there any way to have Asterisk answer from the IP address used instead of using the default one? Regards, 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
Re: [asterisk-users] How to have Asterisk respond from the IP addressused for registration
I should have mentionned this is already done. I can see that is a SIP response when trying 192.168.1.3, but the phones fails to register. I suspect a NAT/firewall issue because packets are leaving for 192.168.1.3, but coming back from 192.168.1.2. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 27, 2010 16:15 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How to have Asterisk respond from the IP addressused for registration 2 things to try - (1) set bindaddr in sip.conf to 0.0.0.0 instead of 192.168.1.2 - in theory this will let * use both cards (2) start second instance of asterisk bound to 192.168.1.3 - probably the approach with the better chance of success. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, May 27, 2010 2:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] How to have Asterisk respond from the IP addressused for registration Hi, I have a test server with 2 NICs, each with it own IP address. Let`s say 192.168.1.2 and 192.168.1.3. I would like some phones to register by using 192.168.1.2 and some by using 192.168.1.3 as the address. Since the default IP is 192.168.1.2, that is the only working address. Every phone connecting to 192.168.1.3 fails to register, presumably because Asterisk answers back from 192.168.1.2 and the phone doesn't recognize this as the correct SIP server. I am using 1.4.31. Is there any way to have Asterisk answer from the IP address used instead of using the default one? Regards, 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
[asterisk-users] Pattern matching - how to ignore numbers after 10 digits
All:
Yesterday I discovered something interesting. I dialed 1800ANCESTRY
from the asterisk system I am testing and got the number doesn't exist
message. I then dialed the same number from our old system and it went
through.
I realized that the Y in ancestry made the number too long, and went
back to my dialplan.
How do I ignore numbers that are too long? Obviously, I've done
something wrong in my pattern matching.
outgoing part of extensions.conf
exten = _91XX!,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net) ; long distance
exten = _9765XXX,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net) ; local
exten = _9XXX,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net) ; local
exten = _9011XXX!,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net) ; international
exten = _911,1,DIAL(SIP/${ext...@ia.ntelos.net) ; emergency
Thanks!
Eddie Mikell
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[asterisk-users] Aastra i740 and Asterisk
Hi listers! Just ran across a customer who wants to replace an Aastra Nexspan with an Asterisk 1.6.X, wants also to connect it to a MOCS (Microsoft Office Comunications Server) though that's not my real concern right now. I got one of his phones (Aastra conexity i740) and though I have been able to change te phone's IP, GW and mask parameters, have not yet a clue on how to make it register with asterisk. Has anyone out there got some experience dealing with something similar?? Thanks! Alyed -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pattern matching - how to ignore numbers after 10 digits
I guess it's the !, sometimes it has a funny behaviour.
try changing (. instead of ! and an X less)
exten =
_91XX!,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.netexten%3a1...@ia.ntelos.net)
; long distance
to
exten =
_91X.,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.netexten%3a1...@ia.ntelos.net)
; long distance
I always use . and never had a problem.
Alyed
2010/5/27 Eddie Mikell ed...@rimmkaufman.com
All:
Yesterday I discovered something interesting. I dialed 1800ANCESTRY
from the asterisk system I am testing and got the number doesn't exist
message. I then dialed the same number from our old system and it went
through.
I realized that the Y in ancestry made the number too long, and went
back to my dialplan.
How do I ignore numbers that are too long? Obviously, I've done
something wrong in my pattern matching.
outgoing part of extensions.conf
exten =
_91XX!,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.netexten%3a1...@ia.ntelos.net)
; long distance
exten =
_9765XXX,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.netexten%3a1...@ia.ntelos.net)
; local
exten =
_9XXX,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.netexten%3a1...@ia.ntelos.net)
; local
exten =
_9011XXX!,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.netexten%3a1...@ia.ntelos.net)
; international
exten = _911,1,DIAL(SIP/${ext...@ia.ntelos.net exten...@ia.ntelos.net)
; emergency
Thanks!
Eddie Mikell
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Re: [asterisk-users] OpenVox B200P and D410P under Asterisk 1.6
Hi! Question, should I be using mISDN or libpri for these cards when they are in the same system, or does DAHDI now support both cards under asterisk 1.6 reliably? I cannot answer that question, but do stay away from mISDN if you can. Philipp -- _ -- 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] Pattern matching - how to ignore numbers after 10 digits
On Thu, 27 May 2010, Eddie Mikell wrote:
exten = _911,1,DIAL(SIP/${ext...@ia.ntelos.net) ; emergency
Unrelated to your question, but 911 doesn't need an underscore.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Meetmee user introduction disabled
David Backeberg wrote: On Thu, May 27, 2010 at 4:05 AM, Theo Band theo.b...@greenpeak.com wrote: First I noted that dahdi_dummy is no longer present in kmod-dahdi-linux-2.3.0.1-1. Not exactly true. myhost01 asterisk # lsmod | grep dahdi dahdi_dummy 5812 0 dahdi_transcode 8968 1 wctc4xxp dahdi_voicebus 42048 2 wctdm24xxp,wcte12xp dahdi 198992 24 dahdi_dummy,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 4096 2 wctdm24xxp,dahdi myhost01 asterisk # dmesg | grep dahdi dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.3.0 What I mean is that it is no longer present in the package: rpm -qf /lib/modules/2.6.18-164.11.1.el5/dahdi/dahdi_dummy.ko kmod-dahdi-linux-2.2.1-1_centos5.2.6.18_164.11.1.el5 rpm -ql kmod-dahdi-linux-2.3.0.1-1_centos5.2.6.18_194.3.1.el5|grep dahdi_dummy.ko Reverting back to kmod-dahdi-linux-2.2.1-1 solved that issue, now the module is loaded again. I suppose it would. I got dahdi_dummy with 2.3.0 by analyzing how the build process worked, and doing some tricks. I could see dahdi_dummy.c was in the package but it wasn't getting built. Here's the trick. If you pull down the combined dahdi package, extract it, cd into the extracted top-level folder cd linux (which is the dahdi proper stuff) make MODULES_EXTRA=dahdi_dummy That worked for me. Do the make install too. asktest01 linux # make MODULES_EXTRA=dahdi_dummy I used to build Asterisk from source including the zaptel-dummy module. Last year I decided to upgrade and use a yum repository. I hoped that this would be less hassle compared to manually chasing after the latest release, compiling etc. And after every kernel update the modules need to be recompiled. The yum flow does it all for me using this repository: [asterisk-current] name=CentOS-$releasever - Asterisk - Current baseurl=http://packages.asterisk.org/centos/$releasever/current/$basearch/ enabled=1 gpgcheck=0 #gpgkey=http://packages.asterisk.org/RPM-GPG-KEY-Digium This is why I prefer not to compile and install anything outside of yum/rpm. It seems that these days you need to provide extra arguments to get dahdi_dummy, and it's getting filtered by default. So what you describe is probably what the package builders also need to know. How can I report such an issue other than using this forum? I don't think a private mail to the maintainer is the best option. And although kmod-dahdi-linux-2.2.1-1 contains the dummy module, the newer version of meetme still does not work. So that's a meetme application issue I think. -- _ -- 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 have Asterisk respond from the IP address used for registration
On Thu, 27 May 2010, Mike wrote: Hi, I have a test server with 2 NICs, each with it own IP address. Let`s say 192.168.1.2 and 192.168.1.3. I would like some phones to register by using 192.168.1.2 and some by using 192.168.1.3 as the address. Since the default IP is 192.168.1.2, that is the only working address. Every phone connecting to 192.168.1.3 fails to register, presumably because Asterisk answers back from 192.168.1.2 and the phone doesn't recognize this as the correct SIP server. I am using 1.4.31. Is there any way to have Asterisk answer from the IP address used instead of using the default one? I think you should take a step back and ask yourself why you are trying to do this in the first place. Presumably you have both of these NIC's plugged into the same logical LAN or you will have even more difficulties with routing later. What problem are you actually trying to solve? 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] [X100P+Dahdi 2.3.0] Couple of questions
Hello, From www.x100p.com, I bought one of those cheap FXO cards. I have a couple of questions/issues about it: 1. I noticed that... - after cold booting the host, I see successful Dahdi/wcfxo messages in /var/log/messages - then, if I run either /etc/init.d/dahdi restart, or /etc/init.d/dahdi stop; /etc/init.d/dahdi start without waiting more than about 10 seconds between the stop/start commands, I get the familiar error messages DAHDI_CHANCONFIG failed on channel 1: No such device or address (6), Failed to initailize DAA, giving up error, and massive FXO PCI Master abort errors in /var/log/messages. According to this thread, this error with X100P cards can be due to some strange wiring: https://issues.asterisk.org/view.php?id=14232 http://www.mail-archive.com/asterisk-...@lists.digium.com/msg35317.html However, this occured on a host running Dahdi 2.3.0: Does it mean that this fix hasn't been ported from Zaptel to Dahdi, or that this error can have another cause? Could it be some timing issue in hardware and/or software, or maybe some initialization issue? In which case, is there a solution? IOW (and I don't mean this as criticism), is the X10xP hardware really crappy by design, or is the real cause for those problems to be found in the Zaptel code which were never really looked into because (understandably) developers prefered to work on the wctdm driver for the more professional TDM cards? 2. This card has the Silicon Labs Si3014/Si3034 chips which are supposed to support global line standards. I'm located in continental Europe, and apparently, for call-progress detection to have any chance to work correctly, I need to change the DAA from FCC (North America) to CTR21 (Europe). Does someone know how to do this? Thank you. -- _ -- 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] Adding a context from the console
On Thursday 27 May 2010 14:05:20 Danny Nicholas wrote: I assume this patch is for 1.6X since I find no code similar to this in 1.4.30? You assume incorrectly. You probably missed that this patch is against pbx_config.c, not main/pbx.c. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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 have Asterisk respond from the IP address used for registration
That was a simplified example. I actually have two links from different ISPs, totally different networks. Those on provider A should talk to provider`s A IP address and have their answers come back from provider's A IP, and those on provider B should talk to my provider B NIC and get the response back from that IP. This is all to make sure latency is kept to a minimum. Provider A`s network doesn't peer to provider B, so latency is horrible when the customer doesn't use the right IP address. The reason why I am not using a diff server per provider is that for some customers, half the phone will be on provider A and half on provider B (home-based personnel). They still keep hints and stuf to work as if they were on the same server. So my original question I believe is still valid, even if the IPs used as exemple made little common sense (as you`ve rightly noted) Michael -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Thursday, May 27, 2010 19:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to have Asterisk respond from the IP address used for registration On Thu, 27 May 2010, Mike wrote: Hi, I have a test server with 2 NICs, each with it own IP address. Let`s say 192.168.1.2 and 192.168.1.3. I would like some phones to register by using 192.168.1.2 and some by using 192.168.1.3 as the address. Since the default IP is 192.168.1.2, that is the only working address. Every phone connecting to 192.168.1.3 fails to register, presumably because Asterisk answers back from 192.168.1.2 and the phone doesn't recognize this as the correct SIP server. I am using 1.4.31. Is there any way to have Asterisk answer from the IP address used instead of using the default one? I think you should take a step back and ask yourself why you are trying to do this in the first place. Presumably you have both of these NIC's plugged into the same logical LAN or you will have even more difficulties with routing later. What problem are you actually trying to solve? 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 -- _ -- 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] pri show version still shows old version despite doing a make make clean make install for v1.4.11
Hi Guys, I am running a PBXinaFLASH server. I replaced contents of /usr/src/libpri with the new version of Libpri v1.4.11. The installed one was v1.4.10. System is running Asterisk 1.4.21.2. I did the following after: cd /usr/src/libpri/ make make clean make install Install end with these lines.: *ln -sf libpri.so.1.4 libpri.so* *mkdir -p /usr/lib* *mkdir -p /usr/include* *install -m 644 libpri.h /usr/include* *install -m 755 libpri.so.1.4 /usr/lib* *#if [ -x /usr/sbin/sestatus ] ( /usr/sbin/sestatus | grep SELinux status: | grep -q enabled); then /sbin/restorecon -v /usr/lib/libpri.so.1.4; fi* *( cd /usr/lib ; ln -sf libpri.so.1.4 libpri.so)* *install -m 644 libpri.a /usr/lib* *if test $(id -u) = 0; then /sbin/ldconfig -n /usr/lib; fi* Is this ^ installed properly? Don't I get a successful message? And here is the result: *r...@pbx:/usr/src/libpri $ asterisk -rx pri show version* *libpri version: 1.4.10.2* What am I doing wrong that it's not update to 1.4.11? 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] How to have Asterisk respond from the IP address used for registration
On 28/05/2010, Mike l...@virtutel.ca wrote: That was a simplified example. I actually have two links from different ISPs, totally different networks. Those on provider A should talk to provider`s A IP address and have their answers come back from provider's A IP, and those on provider B should talk to my provider B NIC and get the response back from that IP. I think this is more a router issue - we do this with three links, going into a single Linux-based Linksys which acts as the single gateway for the LAN (so it has 4 interfaces). You need to look into the ip command, and packet mangling to mark connections as coming from each provider (so that all related packets go back the same way). HTH Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett -- _ -- 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] OpenVox B200P and D410P under Asterisk 1.6
On 5/27/2010 2:33 PM, Philipp von Klitzing wrote: Hi! Question, should I be using mISDN or libpri for these cards when they are in the same system, or does DAHDI now support both cards under asterisk 1.6 reliably? I cannot answer that question, but do stay away from mISDN if you can. Philipp OK thanks Philipp. OpenVox has been steering me toward mISDN for their B200P card, but I am reluctant given what I've learned so far. My experience is only with bristuff, but I had hoped to use the generic DAHDI. -Scott -- _ -- 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] pri show version still shows old version despite doing a make make clean make install for v1.4.11
- bruce bruce bruceb...@gmail.com wrote: What am I doing wrong that it's not update to 1.4.11? Thanks, Bruce -- Did you restart your services to ensure the new library was picked up? --Tim -- _ -- 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] About Sangoma cards and Asterisk integration with other PBX
I suspect the channel is not ceased correctly in Siemens PBX, since you get
dial tone from Siemens PBX the channel from Asterisk is rejected in your
Siemens PBX.
On Thu, May 27, 2010 at 6:15 AM, Daniel Bareiro daniel-lis...@gmx.netwrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Fri, May 21, 2010 at 23:35:14 -0300, Tim Nelson wrote:
Greetings!
Hi, Tim!
I had the opportunity to test a Sangoma A200 card and I have some
doubts that I would like to consult:
I tried to detect the card and I had no success using the wctdm
module with DAHDI. I guess this is because electronics is different
because the TDM400P and OpenVox A400P cards have separate modules for
each channel, while the Sangoma A200 each module operates two
channels. I had to compile Wanpipe for the card was detected. Is it
the only way?
Wanpipe is what interfaces the hardware with Dahdi/Zaptel. Then,
Dahdi/Zaptel interfaces with Asterisk. This is normal.
Well, then wanpipe is necessary.
Another thing I want to try is to connect Asterisk with Siemens PBX
so that the extensions on Asterisk can communicate with the
extensions on the Siemens PBX and vice versa. For this should I
connect a FXO channel on Asterisk with a FXS channel of Siemens PBX?
Personally, if possible, I'd connect one of each(FXO/FXS) on Asterisk
to one of each(FXO/FXS) on the Siemens. This allows for proper dialing
between systems and passing your ${EXTEN} as expected.
I'm not sure if I understood well. Must I use two FXO/FXS connections? A
FXO (Asterisk) / FXS (Siemens) connection and another FXO (Siemens) /
FXS (Asterisk) connection? does not serve a single connection for
incoming and outgoing calls like when we connect Asterisk to the PSTN?
I noticed that, unlike OpenVox A400P card, RJ connectors on the
Sangoma A200 card are smaller. Apparently, the OpenVox use standard
telephone connectors.
Sangoma's cards come with a half-height PCI bracket for smaller
systems. To ensure the card stays small, they use smaller jacks, RJ14
or 'handset' jacks IIRC. Again, this is something specific to Sangoma
and normal.
Today I was doing tests connecting FXO channel on Sangoma card to a
extension of Siemens PBX. Previously, connecting a phone, I made sure in
that socket I had a dial tone.
I tried calling the extension 509 on Siemens PBX, but I get a busy tone
with the following message in the CLI:
- -
dynatac*CLI
-- Executing [9...@from-internal:1] Dial(SIP/200-0004,
DAHDI/3/509) in new stack
[May 26 14:47:59] WARNING[3031]: app_dial.c:1298 dial_exec_full: Unable
to create channel of type 'DAHDI' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [9...@from-internal:2] Hangup(SIP/200-0004, )
in new stack
== Spawn extension (from-internal, 9509, 2) exited non-zero on
'SIP/200-0004'
-- Executing [9...@from-internal:1] Dial(SIP/200-0005,
DAHDI/3/509) in new stack
[May 26 14:48:32] WARNING[3032]: app_dial.c:1298 dial_exec_full: Unable
to create channel of type 'DAHDI' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [9...@from-internal:2] Hangup(SIP/200-0005, )
in new stack
== Spawn extension (from-internal, 9509, 2) exited non-zero on
'SIP/200-0005'
- -
This is the configuration I'm using in chan_dahdi.conf:
- -
[trunkgroups]
[channels]
language=es
defaultzone=es
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
inmediate=no
; DGB - 20100322
busydetect=yes
busycount=3
;Sangoma AFT-A200 [slot:8 bus:1 span:1] wanpipe1
context=from-internal
mailbox=...@voicemail
callerid=Jane Doe 300
group=1
echocancel=yes
signalling = fxo_ls
channel = 1
context=from-internal
group=2
echocancel=yes
signalling = fxo_ks
channel = 2
context=from-zaptel
group=3
echocancel=yes
signalling = fxs_ks
channel = 3
context=from-zaptel
group=4
echocancel=yes
signalling = fxs_ks
channel = 4
- -
And the extensions.conf file is the following:
- -
; DGB - 20100511
[general]
autofallthrough=no
[macro-dial]
exten = s,1,Dial(${ARG1},15)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${macro_ext...@voicemail,u)
exten = s-NOANSWER,n,Hangup
exten = s-BUSY,1,Voicemail(${macro_ext...@voicemail,b)
exten = s-BUSY,n,Hangup
exten = s-CHANUNAVAIL,1,Playback(pbx-invalid)
[from-internal]
; SIP extensions
exten = _2xx,1,Macro(dial,SIP/${EXTEN})
exten = _2xx,n,Hangup
; analog extension
exten = 300,1,Macro(dial,DAHDI/1)
exten
[asterisk-users] Asterisk-based Incredible PBX
Hi all, Today at 12 Noon EDT (9AM PDT, 5PM UK, 6PM Western Europe) the VUC welcomes Ward Mundy from http://NerdVittles.com who will introduce us to Incredible PBX, an Asterisk-based, easy-to-deploy PBX. Rather than start a long chain of features here,we invite you to join us live (see below) or download the recorded version Saturday on the site or via iTunes. To hear the VUC and preferably to join in the discussion with the friendly gang of VoIP/Asterisk enthusiasts (avoiding the g word) see http://vuc.me sip:200...@login.zipdx.com skype:vuc.me irc:freenode.net #vuc (or use the Freenode web client: http://vuc.me/irc ) pstn: +1 567 252 2286 iNum: +883 5100 123 94882 Join us any Friday. If you're at AMOOCON next Friday, I hope to meet you there! /r -- _ -- 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] call-waiting
hi, all Is ther any way to set up call-waiting feature in asterisk using dialplan or any other ways. I want to use only asterisk for that not any other gui. I am using asterisk 1.4.28. Regards, -- Bhrugu Mehta Sr. S/W Engineer (DD) VOIP,Telephony Team India -- _ -- 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
