Re: [asterisk-users] Enable/Disable Sip without registration
Have a look here: http://www.voip-info.org/wiki-Asterisk+config+sip.conf and read: http://www.h6315.com/ast_docs/Asterisk%20TFOT%20v2.pdf - Original Message - From: equis software To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, December 12, 2007 4:05 PM Subject: Re: [asterisk-users] Enable/Disable Sip without registration Sorry I don´t understand. Could you explain me with more detailed? Thanks! On Dec 12, 2007 10:35 AM, ram <[EMAIL PROTECTED]> wrote: On Dec 12, 2007 6:31 PM, equis software <[EMAIL PROTECTED]> wrote: I try to configure that only registered sips can make calls. How can I do that? I was looking in sip.conf but I didn´t found wath opition configure this functionality. Create a users in sip.conf with context so that user will register with asterisk to make calls ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.278 RTP conversation capture, please.
Why don't you run tcpdump on any SIP server ? (Or are you emailing here because you don't have one and need one ? If that is the case can I ask why you need it ?) - Original Message - From: Kerry S To: asterisk-users@lists.digium.com Sent: Wednesday, December 19, 2007 2:11 AM Subject: [asterisk-users] G.278 RTP conversation capture, please. Hello all, I have a bit of a request. I need a wireshark capture of a SIP conversation using g.728. I don't need anything fancy, just a call and have both ends say "hi" to each other. hopefully someone out there can help me. Thank you all. This list has been of use many times in the past, even though I tend to stay quiet. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PXE-bootable diskless Asterix distro?
On Sat, 22 Dec 2007 21:01:47 -0600, "Michael Graves" <[EMAIL PROTECTED]> wrote: >I'm not at all certain what you need to change on the hardware, but it >seems to me it should be trivial. Perhaps something in the BIOS? I was looking at ways to boot it up over the network, and keep everything in RAM, but maybe I'll be lucky and it'll start over a USB key. The BIOS is supposed to date from "10081999". I'm still not clear about the difference between the two BIOSes "Workstation on demande" and "Netstation manager". There's a very long discussion on installling DamnSmallLinux on that hardware: http://damnsmalllinux.org/cgi-bin/forums/ikonboard.cgi?act=ST;f=12;t=3204;st=10 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PXE-bootable diskless Asterix distro?
On Sun, 23 Dec 2007 06:15:22 +0200, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: >Yes. I have a version of our CD that boots from PXE. It took minor >changes and rebuilding as a "PXE image", as Debian Live has basic >support of that already. For simplicity I figure you'll be after a >system that has everything in the initrd, but this is not the case here. >It mounts a network partition to do the rest. We use NFS. CIFS is also >supported. Since ultimately the whole system will have to run from the CF card, I'm looking for something that can be downloaded from a remote server through PXE, and then run entirely from RAM. In the mean time, provided the hardware can boot off a Debian Live CD, it's fine: I just need to check that the Netvista can run Asterisk and the TDM card with good performance before bothering further. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PXE-bootable diskless Asterix distro?
On Sun, Dec 23, 2007 at 09:06:17AM +0100, Vincent wrote: > On Sun, 23 Dec 2007 06:15:22 +0200, Tzafrir Cohen > <[EMAIL PROTECTED]> wrote: > >Yes. I have a version of our CD that boots from PXE. It took minor > >changes and rebuilding as a "PXE image", as Debian Live has basic > >support of that already. For simplicity I figure you'll be after a > >system that has everything in the initrd, but this is not the case here. > >It mounts a network partition to do the rest. We use NFS. CIFS is also > >supported. > > Since ultimately the whole system will have to run from the CF card, > I'm looking for something that can be downloaded from a remote server > through PXE, and then run entirely from RAM. > > In the mean time, provided the hardware can boot off a Debian Live CD, > it's fine: I just need to check that the Netvista can run Asterisk and > the TDM card with good performance before bothering further. Debian Live has several targets: * iso: a standard bootable CD image. * usb-hdd: An image to copy (with dd) to a USB storage device * net: files for booting the distribution through PXE. We currently build just the "iso" target on what we ship outside. the "net" target mostly uses the same tree. And I have a separate PXE setting here, so I don't need the full tree. I think that the contents of the CD could be tweaked to be booted from PXE with very minimal changes. But I can't think of giving instructions. All the PXE instructions pages I saw to date assume that $DISTRO is the only thing you're going to boot from PXE. Creating a PXE boot menu to allow booting several of them is rather trivial, but not well-documented. If you want to try this I can assist you in private mail. Or you can try to tweak it yourself, using a standard Debian / Ubuntu as a build system and the config from the CD as a reference. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.26 badly broken?
After upgrading from 1.2.25 to 1.2.26 i noticed that IAX -> IAX calls always result in Asterisk just exiting without any message. Asterisk also seems to die when using a TDM400 with 2 FXO modules, placing 2 outgoing calls on both lines as Zap/g2 and then trying to make a 3rd call. Went back to 1.2.25 for the moment ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.26 badly broken?
On 11:51, Sun 23 Dec 07, Remco Barendse wrote: > After upgrading from 1.2.25 to 1.2.26 i noticed that IAX -> IAX calls > always result in Asterisk just exiting without any message. > > Asterisk also seems to die when using a TDM400 with 2 FXO modules, placing > 2 outgoing calls on both lines as Zap/g2 and then trying to make a 3rd > call. > > Went back to 1.2.25 for the moment For this very reason they released 1.2.26.1 Also note that 1.2 is no longer maintained. It gets security fixes only. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD "Why is it drug addicts and computer afficionados are both called users?" ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP handling - why 491 Request Pending response
Olle, You're right. I missed one thing when I concluded that this was an INVITE glare condition, which is that when the UAC and UAS dialogs are matched they're compared with respect to their LOCAL and REMOTE tags as opposed to the To and From tags. The LOCAL and REMOTE tags will get filled either with To or From tag depending on whether you're looking at the dialog from the UAC or from the UAS. So, in this case you'll have two dialogs in Asterisk and they'll have the same Call-Id but their tags will be swapped. Do you think Asterisk dialog matching is coded to handle this? While we've not seen a trace yet, it'd seem like we're missing something because we're sending back a 491. Raj On Dec 23, 2007 2:21 AM, Johansson Olle E <[EMAIL PROTECTED]> wrote: > > 23 dec 2007 kl. 01.45 skrev Raj Jain: > > > You can not do this. You can not have an INVITE that Asterisk > > originated enter back into Asterisk. Technically this is not a loop, > > but this is an INVITE glare and the way Asterisk is reacting is > > correct. > > > > You'll need to change the Call-Id of the INVITE that goes into > > Asterisk (a proxy can not do that so you'll need a B2BUA), or else > > you can do something like what Olle suggested. > > I don't really agree here Raj. Of course you can send an INVITE to an > URI hosted by the proxy and the location table points back to one or > several URI's in the same Asterisk server. > > /O > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need some one to make a test call
Hi, I am looking for some one to make a test call for me to a toll free number in Australia (from a land line in Australia) and to a toll free number in Argentina (from land line in Argentina). The numbers are set to echo test at the moment. Thanks. /J ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP handling - why 491 Request Pending response
it seems that asterisk in not cheking tags in to and from headers. When asterisk responds to the second INVITE it puts again the same tag in the To header: tag=as62247c57 what kind of trace can I catch to have more details here? On Dec 23, 2007 12:51 PM, Raj Jain <[EMAIL PROTECTED]> wrote: > Olle, > > You're right. I missed one thing when I concluded that this was an INVITE > glare condition, which is that when the UAC and UAS dialogs are matched > they're compared with respect to their LOCAL and REMOTE tags as opposed to > the To and From tags. The LOCAL and REMOTE tags will get filled either with > To or From tag depending on whether you're looking at the dialog from the > UAC or from the UAS. So, in this case you'll have two dialogs in Asterisk > and they'll have the same Call-Id but their tags will be swapped. > > Do you think Asterisk dialog matching is coded to handle this? While we've > not seen a trace yet, it'd seem like we're missing something because we're > sending back a 491. > > Raj > > > > On Dec 23, 2007 2:21 AM, Johansson Olle E <[EMAIL PROTECTED]> wrote: > > > > > 23 dec 2007 kl. 01.45 skrev Raj Jain: > > > > > You can not do this. You can not have an INVITE that Asterisk > > > originated enter back into Asterisk. Technically this is not a loop, > > > but this is an INVITE glare and the way Asterisk is reacting is > > > correct. > > > > > > You'll need to change the Call-Id of the INVITE that goes into > > > Asterisk (a proxy can not do that so you'll need a B2BUA), or else > > > you can do something like what Olle suggested. > > > > I don't really agree here Raj. Of course you can send an INVITE to an > > URI hosted by the proxy and the location table points back to one or > > several URI's in the same Asterisk server. > > > > /O > > > > ___ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > 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-- > > 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP handling - why 491 Request Pending response
You may want to try setting pedantic=yes in your sip.conf and retest this. The following traces would be helpful: sip set debug on core set debug 10 core set verbose 10 You may want to open a bug in bug tracker and upload your config and log files over there. -- Raj On Dec 23, 2007 7:47 AM, Tomasz Zieleniewski <[EMAIL PROTECTED]> wrote: > it seems that asterisk in not cheking tags in to and from headers. > When asterisk responds to the second INVITE it puts again the same tag in > the To header: tag=as62247c57 > > what kind of trace can I catch to have more details here? > > > On Dec 23, 2007 12:51 PM, Raj Jain <[EMAIL PROTECTED]> wrote: > > > Olle, > > > > You're right. I missed one thing when I concluded that this was an > > INVITE glare condition, which is that when the UAC and UAS dialogs are > > matched they're compared with respect to their LOCAL and REMOTE tags as > > opposed to the To and From tags. The LOCAL and REMOTE tags will get filled > > either with To or From tag depending on whether you're looking at the dialog > > from the UAC or from the UAS. So, in this case you'll have two dialogs in > > Asterisk and they'll have the same Call-Id but their tags will be swapped. > > > > Do you think Asterisk dialog matching is coded to handle this? While > > we've not seen a trace yet, it'd seem like we're missing something because > > we're sending back a 491. > > > > Raj > > > > > > > > On Dec 23, 2007 2:21 AM, Johansson Olle E <[EMAIL PROTECTED]> wrote: > > > > > > > > 23 dec 2007 kl. 01.45 skrev Raj Jain: > > > > > > > You can not do this. You can not have an INVITE that Asterisk > > > > originated enter back into Asterisk. Technically this is not a loop, > > > > but this is an INVITE glare and the way Asterisk is reacting is > > > > correct. > > > > > > > > You'll need to change the Call-Id of the INVITE that goes into > > > > Asterisk (a proxy can not do that so you'll need a B2BUA), or else > > > > you can do something like what Olle suggested. > > > > > > I don't really agree here Raj. Of course you can send an INVITE to an > > > URI hosted by the proxy and the location table points back to one or > > > several URI's in the same Asterisk server. > > > > > > /O > > > > > > ___ > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > > > 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-- > > > > 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-- > > 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enable/Disable Sip without registration
On Dec 12, 2007 8:01 AM, equis software <[EMAIL PROTECTED]> wrote: > I try to configure that only registered sips can make calls. > How can I do that? Registrations are meant for routing calls to end-points, not for accepting calls from end-points. I don't think Asterisk supports a mechanism which allows only registered end-points to make calls. -- Raj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PXE-bootable diskless Asterix distro?
On Sun, 2007-12-23 at 09:06 +0100, Vincent wrote: > On Sun, 23 Dec 2007 06:15:22 +0200, Tzafrir Cohen > <[EMAIL PROTECTED]> wrote: > >Yes. I have a version of our CD that boots from PXE. It took minor > >changes and rebuilding as a "PXE image", as Debian Live has basic > >support of that already. For simplicity I figure you'll be after a > >system that has everything in the initrd, but this is not the case here. > >It mounts a network partition to do the rest. We use NFS. CIFS is also > >supported. > > Since ultimately the whole system will have to run from the CF card, > I'm looking for something that can be downloaded from a remote server > through PXE, and then run entirely from RAM. > Hi Vincent, Seems what you write is somehow misleading A) the whole the have to run from CF. Well fine, as long as the cf-disk is on-its-way, you can emulate it with an usb-stick/pen-drive. No neet for PXE anyway.. B) I'm looking for something that can be downloaded from a remote server through PXE, and then run entirely from RAM. Also fine, but why are you then waiting for the CF? is this: http://www.automated.it/asterisk/ perhaps something for you? hw ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PXE-bootable diskless Asterix distro?
On Sun, Dec 23, 2007 at 03:30:50PM +0100, Hans Witvliet wrote: > On Sun, 2007-12-23 at 09:06 +0100, Vincent wrote: > > On Sun, 23 Dec 2007 06:15:22 +0200, Tzafrir Cohen > > <[EMAIL PROTECTED]> wrote: > > >Yes. I have a version of our CD that boots from PXE. It took minor > > >changes and rebuilding as a "PXE image", as Debian Live has basic > > >support of that already. For simplicity I figure you'll be after a > > >system that has everything in the initrd, but this is not the case here. > > >It mounts a network partition to do the rest. We use NFS. CIFS is also > > >supported. > > > > Since ultimately the whole system will have to run from the CF card, > > I'm looking for something that can be downloaded from a remote server > > through PXE, and then run entirely from RAM. > > > > Hi Vincent, > Seems what you write is somehow misleading > A) the whole the have to run from CF. > Well fine, as long as the cf-disk is on-its-way, you can emulate it with > an usb-stick/pen-drive. No neet for PXE anyway.. > > B) I'm looking for something that can be downloaded from a remote server > through PXE, and then run entirely from RAM. > Also fine, but why are you then waiting for the CF? Run entirely from RAM? So if your image is 40MB, you waste 40MB of RAM just to store it? This is where (A) becomes suddenly a lot nicer. Especially if you can use union-mouting and thus to have to use such a specific system. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PXE-bootable diskless Asterix distro?
On Sun, Dec 23, 2007 at 04:41:42PM +0200, Tzafrir Cohen wrote: > Run entirely from RAM? So if your image is 40MB, you waste 40MB of RAM > just to store it? This is where (A) becomes suddenly a lot nicer. > Especially if you can use union-mouting and thus to have to use such a > specific system. 40MB is a paltry amount of RAM for any even reasonably modern machine. Compare, too, the respective access times between flash and RAM, not to mention the write session limits of flash (though again, to consider this is to make another mountain out of a mole hill). The only significant downside of using a net-boot setup (e.g. PXE) for something like this is that you are then reliant on another machine for the operation of that one. Considering that you'd probably be relying on it (and the switch) for normal operations anyway, I would not bother worrying about it. -- Ulexus [EMAIL PROTECTED] pgpwb6J2T8aYm.pgp Description: PGP signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
hi: thanks a lot for so many great information. i tried to read the specs and manuals for all the phones mentioned. we use alcatel pbx in most offices. i surveyed some users to understand what functions they use most. and i found few people know how to use 3way-conf or forward.i think if the function needs two or more keys to operate, then people tend to ignore it unless he use that function for daily business. i conclude the functions we need are all basic functions. but due to the difference of ip pbx/phones and classic pbx/phones, some of these functions seem not so "basic" in the ip world: 1. dial out name display. when you dial a number, the phone lcd will show the corresponding name, so you can realize if it is the correct number immediately. this needs a corporate directory support, or put the whole corporate phonebooks to every ip phone. most ip phone has less than 500 local phonebook entries. this is not enough for us. grandstream: has xml phonebook support and can combine with local phonebooks. linksys: has coporate directory but seems only work with linksys pbx, not asterisk. aastra: has xml phonebook snom: has ldap and xml phonebook. xml seems for browsing,don't know if work here. other china brand phone: none. 2. transfer. transfer is simple and straightforward in classic pbx. you just press "transfer" then dial number and you are on the way of attended transfer. you press "transfer" again to cancel transfer. you hangup to complete the attended transfer. if you hangup before the completion of attended transfer, the transfer becomes blind transfer automatically. eventually user didn't notice the "blind" or "attended" concept in classic pbx. snom: has "transfer on hook". don't know if it can do all what i want. others: some china phones almost can do it, but need to press "hold" to cancel transfer. 3. call back on busy. in alcatel, if you dial someone and he is on the phone, you will hear something like "busy, please dial 5 if you want to request callback". you can dial 5 and you will hear "success, please hangup". asterisk has several ways and patches to do this. but i saw some phone can do this locally. i don't know which is better. linksys: has this function in spec. don't know how to use. snom: has "call completion". others: i didn't find this or i miss it. 4. pickup. i think this is easy to emulate "*8" and let asterisk do it. any better method? every phones can do this emulation. 5. three-way conference, forward. if there are simple (one key) method to implement these. in alcatel, if the phone if forwarded, when you pick up the handset you will hear like "forwarded, please press *1 to cancel". it's easy so everyone can cancel the forward. but it need two keys to start a forward, so few users know how to forward a number. please correct me if there are mistakes or missing. thanks again for your great help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed
Hi i've got an openvox a800p01 with 1 FXO and 4 FSX i've done the following: - downloaded zaptel-1.4.7.1 > >> - downloaded the file opvxa1200.c > >> - copied in zaptel-1.4.7.1/ > >> - edited makefile adding opvxa1200 in the modules and the voice > >> opvxa1200.o : zaptel.h wctdm.h > >> - edited zaptel.sysconfig adding MODULES="$MODULES opvxa1200" # OPENVOXA1200P after that ive done: make clean", "make", "make install finally, if i do: modprobe opvxa1200 if i launch ./zapconf the file /etc/zaptel.conf still remains empty, if i force editing the file adding: > >> fxsks=1 > >> fxoks=2 > >> fxoks=3 > >> fxoks=4 > >> fxoks=5 > >> > >> loadzone= it > >> defaultzone = it and do a: ztcfg - i get: Zaptel Version: 1.4.7.1 > >> Echo Canceller: MG2 > >> Configuration > >> == > >> > >> > >> Channel map: > >> > >> Channel 01: FXS Kewlstart (Default) (Slaves: 01) > >> Channel 02: FXO Kewlstart (Default) (Slaves: 02) > >> Channel 03: FXO Kewlstart (Default) (Slaves: 03) > >> Channel 04: FXO Kewlstart (Default) (Slaves: 04) > >> Channel 05: FXO Kewlstart (Default) (Slaves: 05) > >> > >> 5 channels to configure. > >> > >> ZT_CHANCONFIG failed on channel 1: No such device or address (6) > >> Can you help me to guess the problem? thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed
On Sun, Dec 23, 2007 at 07:05:37PM +0100, nik600 wrote: > Hi > > i've got an openvox a800p01 with 1 FXO and 4 FSX > > i've done the following: > - downloaded zaptel-1.4.7.1 > > >> - downloaded the file opvxa1200.c > > >> - copied in zaptel-1.4.7.1/ > > >> - edited makefile adding opvxa1200 in the modules and the voice > > >> opvxa1200.o : zaptel.h wctdm.h > > >> - edited zaptel.sysconfig adding > MODULES="$MODULES opvxa1200" # OPENVOXA1200P > > after that ive done: > make clean", "make", "make install > finally, if i do: > modprobe opvxa1200 > > if i launch ./zapconf xpp/utils/zapconf ? > > the file /etc/zaptel.conf still remains empty, Which suggests that the module hasn't really loaded or anyway did not register channels. Or it has, but they are for empty slots. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enable/Disable Sip without registration
23 dec 2007 kl. 15.15 skrev Raj Jain: > > On Dec 12, 2007 8:01 AM, equis software <[EMAIL PROTECTED]> > wrote: > I try to configure that only registered sips can make calls. > How can I do that? > > Registrations are meant for routing calls to end-points, not for > accepting calls from end-points. I don't think Asterisk supports a > mechanism which allows only registered end-points to make calls. You are right in that the SIP channel doesn't bother with this. But there's another way. Like most other things related to calls in Asterisk, this can be controlled in the dialplan. Use the SIPPEER function to check if the peer is registered. If not, hangup without answering. If it is, then process the call as usual. If you want to, you can even play prompts or send the calls from unregistered devices to the talking clock in London. Some devices, like Linksys/Sipura, have settings of their own that block calls if the device can't register. Merry Christmas! /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed
On Dec 23, 2007 7:24 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > xpp/utils/zapconf ? yes > > > > > the file /etc/zaptel.conf still remains empty, > > Which suggests that the module hasn't really loaded or anyway did not > register channels. Or it has, but they are for empty slots. > can you suggest me some command to enable the debug of the module? the card has phisically installed the module, i've checked it. And it is correctly powered. Do you know some method to check if the card is working? here the lspci ouput: > 00:0e.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN > interface > Subsystem: Unknown device 9100:0001 > Flags: bus master, medium devsel, latency 32, IRQ 10 > I/O ports at a800 [size=256] > Memory at f800 (32-bit, non-prefetchable) [size=4K] > Capabilities: [40] Power Management version 2 > -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed
On Sun, Dec 23, 2007 at 07:05:37PM +0100, nik600 wrote: > Hi > > i've got an openvox a800p01 with 1 FXO and 4 FSX > > i've done the following: > - downloaded zaptel-1.4.7.1 > > >> - downloaded the file opvxa1200.c > > >> - copied in zaptel-1.4.7.1/ > > >> - edited makefile adding opvxa1200 in the modules and the voice > > >> opvxa1200.o : zaptel.h wctdm.h > > >> - edited zaptel.sysconfig adding > MODULES="$MODULES opvxa1200" # OPENVOXA1200P > > after that ive done: > make clean", "make", "make install > finally, if i do: > modprobe opvxa1200 dmesg | tail > > if i launch ./zapconf cat /proc/zaptel/* -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answering Machine Detection
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tong wrote: > If i use AMD() or the code below, now the problem is the fax machine/modem > detection and answer machine detection get detected as the same. If i need > to seperate the two how do i do that? For example, if i use AMD() to detect > an answer machine by saying any greeting exceeding 2.5 seconds is a machine, > how do i distinguish between a fax/modem and a long greeting? This is a well known problem, which has a patch, which has always fixed the problem for me. Please install the patch, and if it fixes it for you, add to the bugtracker: http://bugs.digium.com/view.php?id=9256 - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHbss7DQNt8rg0Kp4RApW1AJ0eMu4y9Wp2pI83mHKcPre90K8fRQCeLxfK 4shNRHmxmWvhyZmiWrfo7dE= =1nfW -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answering Machine Detection
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tong wrote: > If i use AMD() or the code below, now the problem is the fax machine/modem > detection and answer machine detection get detected as the same. If i need > to seperate the two how do i do that? For example, if i use AMD() to detect > an answer machine by saying any greeting exceeding 2.5 seconds is a machine, > how do i distinguish between a fax/modem and a long greeting? Woops, sorry was responding to an earlier post. The patch before was so that AMD will work. With regard to telling the difference between a fax and a long greeting, you would need to use something like NVFaxDetect (which seems to have disappeared) on VoIP trunks, or you could just turn on fax detection in zapata.conf if you are using Zaptel (and create a fax extension if you want to do something with it). - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHbsvPDQNt8rg0Kp4RAlBEAJ9SU+VDoENogW+evKDl9OJ4J706ggCgr3/z g/gyQZxtc42dc9vn/uWckko= =lf6I -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Active Calls
Hi Friends, Happy New Year I was developing billing system for my end user customers. I need to get Asterisk Active calls in MySQL database with full status of call likem ringing, UP and runtime? i will be thank full for your help and suggestion. Thank You - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PXE-bootable diskless Asterix distro?
On Sun, 23 Dec 2007 15:30:50 +0100, Hans Witvliet <[EMAIL PROTECTED]>
wrote:
>Seems what you write is somehow misleading
I meant that I ordered some CF cards, but until they get here, and
since this baby can boot off a remote server with PXE, I was looking
for a PXEd Asterisk that I could use to check performance :-)
>is this: http://www.automated.it/asterisk/ perhaps something for you?
Exactly. It's pretty old ("This version is built using
asterisk/zaptel/libpri 1.0.7"), but I'll give it a shot. Thanks.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PXE-bootable diskless Asterix distro?
On Sun, 23 Dec 2007 10:10:33 -0500, Ulexus <[EMAIL PROTECTED]> wrote: >Compare, too, the respective access times between flash and RAM, not to >mention the write session limits of flash (though again, to consider >this is to make another mountain out of a mole hill). I'll probably put Linux + Asterisk on the CF card, and let data (WAV files, logs, etc.) live in RAM. A script could be run at shutdown to save data onto the CF card. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to setup redundant SIP peers
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Thomas Balsfulland wrote: > this is not the right way, because it takes 30 sec before asterisk try > dialout over peer2. the dial-timer ([EMAIL PROTECTED],30) is normaly set for > the time to wait of "200 OK", but it is also active when peer1 is > down or not answering. 1st off use qualify to check that peers are up. We also use a script that checks qualify status, and changes a flag in the routing database so that if a provider is down or lagged, they won't be considered in the LCR. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHbvgJDQNt8rg0Kp4RAggUAKC9ujWPzE2iemsMHA4hqyB100McsQCfTJ3+ sZCDlhBzXx+nAuwtAViRRcY= =gQJe -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Soundcard necessary on an asterisk server toget output of playback()??
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Stefan Guenther wrote: > Hi, > > >However, I believe that zaptel >= 1.4.6 or zaptel 1.2 >= 1.2.21 should > >support hires timers for timing on kernel >= 2.6.22 . > > > >What version of Zaptel do you use? > > > I was using version 1.4.5.1 > > I just downloaded and installed version 1.4.7, configure/make/make > install finished without an error, but when is used > > modprobe ztdummy > > the system said: > > FATAL: Error inserting ztdummy > (/lib/modules/2.6.22-14-386/misc/ztdummy.ko): Unknown symbol in module, > or unknown parameter (see dmesg) Are you sure that the source of your kernel is the same as the running kernel? I.E. Have a look at the source it is using while compiling Asterisk and compare that to uname -a - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHbvsnDQNt8rg0Kp4RAoYJAJ9+mTSPlspFuuw4bs0+nT8Rc2EmJgCfbgWB Eo963LjkcgCYMAaVf4NCo8g= =ypV5 -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call center scenario
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jay Moore wrote:
> Greetings, List.
>
> I would like to implement a procedure in my call center but am not sure
> the best way to implement it. I'm hoping I can describe it here and
> that I'll receive some feedback and/or suggestions on how to proceed.
>
> Here's my situation:
>
> My call center fields calls regarding internet access issues for local
> apartment complexes and businesses. Most of the time, we get a few
> calls here and there from new tenants unsure how to set up their
> connection. Every so often, however, there will be some sort of issue
> (ISP going down, router crashing, etc...) that will leave all users
> without internet access. When this happens, we get a flood of calls and
> the girls in my call center can quickly become overwhelmed.
>
> What I'd like to do is set up a system whereby incoming calls during a
> known outage are instead redirected to a recording explaining the issue
> and the option to have the caller leave a message (a la voicemail). All
> calls come down our T1 and we are able to identify the incoming account
> based on its DID. We would need to do this on a per-account basis. My
> girls would also need to have the ability to toggle on/off the
> redirection as well as record a message for the caller to hear -- at a
> moment's notice.
>
> Since my girls only field the calls and don't do any actual support (I
> do that), it'd be ideal if my VM indicator would also let me know if any
> callers left messages during a known outage. Again, this would be
> ideal, but most certainly not necessary.
>
> So, what say you list? Any suggestions on the most efficient way to do
> this? I am quite familiar with PHP and not adverse to writing a script
> to do this for me (I suspect I will have to anyway), but don't wish to
> reinvent the wheel if something like this already exists.
I'd do the same thing we do with customer systems with regard to closing
time.
We give them an extension to turn on forwarding and one to turn off
forwarding.
I.E. 161-179 for various forward locations and 160 for normal operation.
If someone dials 161-179 it sets a value in the Asterisk DB (see
function DB) and if they dial 160 it sets the value to 9.
We then check the value of the DB entry, and if it is 9 we jump to the
standard context. If it is not 9 we continue in the dialplan and send
the call to the value in the field.
You could use the value in the db for an extension to dial.
I.E.
[extensions]
; Turn off forwarding
exten => 160,1,Set(DB(forward/number)=9)
exten => 160,2,Background(call-forward)
exten => 160,3,Background(disabled)
exten => 160,4,Goto(2)
; Forward to Bob Jacks mobile
exten => 162,1,Set(DB(forward/number)=027xxx)
exten => 162,2,Background(call-forward)
exten => 162,3,Background(on)
exten => 162,4,Goto(2)
; Forward to Louise
exten => 166,1,Set(DB(forward/number)=021xxx)
exten => 166,2,Background(call-forward)
exten => 166,3,Background(on)
exten => 166,4,Goto(2)
; Forward to Adrian
exten => 167,1,Set(DB(forward/number)=021xxx)
exten => 167,2,Background(call-forward)
exten => 167,3,Background(on)
exten => 167,4,Goto(2)
[checkforward]
; If the forward number is 9 then dont forward
exten => s,n,GotoIf($["${DB(forward/number)}" = "9"]?678|1:677|1)
; Number was not 9 - lets forward
exten => 677,1,NoOp(Night Time Mode - forward to ${DB(forward/number)})
exten => 677,n,Goto(dialout,${DB(forward/number)},1)
; Number was 9 - accept the call
exten => 678,1,NoOp(Day Time Mode)
exten => 678,n,Goto(daytime,s,1)
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFHbv1wDQNt8rg0Kp4RAkOzAJ4nezHFNGpQirFCKovOcrVe8zthWACfQtzq
dcCbdHNPLQSJYUmdblHOtms=
=KUVl
-END PGP SIGNATURE-
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM and CDMA Gateways
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dovid B wrote: > And whats it called ? Where did you get it from ? > - Original Message - > From: Cameron Hissey > To: Asterisk Users Mailing List - Non-Commercial Discussion > Sent: Friday, December 07, 2007 2:41 AM > Subject: Re: [asterisk-users] GSM and CDMA Gateways > > > i am using a GSM gateway that converts to a PSTN line which i have running > into a TDM400P card. > works a treat! Ignoring top posting - sorry. We use the 2N boxes - which although a little expensive have been great. Every now and then we get a loud sound which was causing the echo canceller in zaptel to go crazy and ended up killing the call. Once I realised that the 2N box was using US impedance instead of NZ impedance, the echo cancelling wasn't required and therefore the problem disappeared. There is a bit of PDD (Post Dial Delay), but this is because of the telco. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHbv+MDQNt8rg0Kp4RAgbZAJ0TYvm+Sw0NLX0u16eIdYJs9X9D6QCeKuax bKcQ7NMPRm7GaiTh4daJoyI= =uUpx -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DeadAgi
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Edwards wrote: > On Thu, 6 Dec 2007, Bhrugu Mehta wrote: > >> I am new to use DeadAgi, >> can anybody help me how to use DeadAgi, >> >> actually i want this, >> >> when caller hangup his/her phone, i want to send packet to my other app that >> check caller hung up done. > > It's "dead" easy :) > > Deadagi() is just like agi() with a couple of differences. In fact (in > 1.2), both are defined in res_agi.c and "dead" is just a flag passed to > run_agi(). > > The differences are how much interaction you can have with the channel* > and whether your AGI gets a "HUP" when the channel is hung up. > > While deadagi() is usually executed in the "h" extension, it is not a > requirement. > > *) I was curious what the exact differences were so I started reading > through ast_waitfor_nandfds(). Unfortunately I have a very low frustration > level for code written with single character variable names like c and n. There is some debate over this but the gist of it is that AGI will send a kill to your script if the channel is hung up. DeadAGI won't. You're now not supposed to run AGI on a channel that dies and not supposed to run DeadAGI on a live channel. We've had to split our billing and routing for this, and since doing so everything has worked fine. Asterisk 1.6 will not suffer this problem and DeadAGI will be deprecated. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHbwAJDQNt8rg0Kp4RAjUzAJ9jq1sFitOR7yG//0GgP5A7lvDbBgCfQg3P n/o4vp6IlSghf2AjNbOFpjU= =Yp/5 -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting custom field in CDR
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Edwards wrote: > On Thu, 6 Dec 2007, Atis Lezdins wrote: > >> Only custom field you can use for now is CDR(userfield) and if you want >> multiple values, you can separate them by something like # > > Adding multiple fields using userfield is a major hack that will > repeatedly bite you for years. > > cdr_addon_mysql.c in the Asterisk addons package works great. It will take > a minor bit of C skills to cut and paste to add new columns. I wrote a patch for Asterisk (which is on the bugtracker) some time ago to add 5 userfields. We still use this on our systems, but the patch on the tracker has grown somewhat stale and will not be included because it was deemed the incorrect way to do it (if you want to do it properly you should allow some custom configuration in cdr_mysql.conf etc). If you need any help applying the patch drop me a line and I'll see what I can do. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHbwDdDQNt8rg0Kp4RAqtSAJ9x6+rEXLa5/MnhjgIHM3mfexXVCACeLPPp yEE7cwdyISAcalJlxbjT6QI= =CBu0 -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Zaptel] Why no port to Windos?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Vincent wrote: > On Fri, 14 Dec 2007 15:47:38 +1100, Paul Hales > <[EMAIL PROTECTED]> wrote: >> Umm - you could just buy a SPA-3000/3102/3666/etc. > > Thanks but I prefer PCI cards. Less cables, less power units that can > burn, less mess :-) It seems strange to make this comment (i.e. higher uptime) in a conversation about porting zaptel to windows. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHbwrtDQNt8rg0Kp4RArqTAJ9Juhkm0UHoFpTkkTl62jj48YPA8gCdEhva +f1tqtO29Ltv7o8zu+raO5U= =7+B9 -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS gateway recommendation
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Robert McNaught wrote: > Im looking to just test the concept of sending SMS texts from *. > > When you say a provider? What kind of provider do you mean? I personally use clickatell. But I use a PHP script to do the texting and call that from the dialplan. I developed a softphone a few years ago that would use the interface too, and would send messages between softphones or cellphones based on the number. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHbwTwDQNt8rg0Kp4RAtZ9AKC4mhvoEZZGsAG0cLXOGtJBzXXKegCePZKm jK2yLqyn0nb71Si67cAsQWE= =Qp0x -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
Hi, Try atcom. www.atcom.com.cn We have tested atcom and its quality also good. they are using infeneon chipset. its support asterisk, sip, iax as well.decent look. cost effetive. still they have basic ip phone modles. starting from next year they will release new modles. Regards, vidura. > > -- > > Message: 1 > Date: Fri, 21 Dec 2007 12:04:57 +0100 > From: Fredrik S?derlund <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] ip phone suggestion for Asia? > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Check out yntx > www.yntx.com > fear prices and recides in Asia and iss it sip on asteriks they will do ! > try to buy one to trye it out before buying fore hole company.. > > /MVH Fille > > > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DeadAgi
On Mon, 24 Dec 2007, Matt Riddell wrote: > There is some debate over this but the gist of it is that AGI will send > a kill to your script if the channel is hung up. DeadAGI won't. res_agi.c says SIGHUP, not SIGKILL. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
hi: thanks for the information. you are the second one who mentioned atcom. so i think this phone has basic quality. i don't have atcom in hand. but i have other china brand(fanvil) phone which seems the same as atcom: infeneon based, sip, iax, good sound quality. but it has poor firmware support and limited function. i check the atcom manual, but didn't find the functions i need (corporate phonebook, transfer, callback..etc). Regards, tbskyd 2007/12/24, Vidura Senadeera <[EMAIL PROTECTED]>: > Hi, > > Try atcom. www.atcom.com.cn > > We have tested atcom and its quality also good. they are using infeneon > chipset. its support asterisk, sip, iax as well.decent look. cost effetive. > still they have basic ip phone modles. starting from next year they will > release new modles. > > Regards, > vidura. > > > > > > > > > -- > > > > Message: 1 > > Date: Fri, 21 Dec 2007 12:04:57 +0100 > > From: Fredrik S?derlund <[EMAIL PROTECTED]> > > Subject: Re: [asterisk-users] ip phone suggestion for Asia? > > To: asterisk-users@lists.digium.com > > Message-ID: <[EMAIL PROTECTED]> > > Content-Type: text/plain; charset="us-ascii" > > > > Check out yntx > > www.yntx.com > > fear prices and recides in Asia and iss it sip on asteriks they will do ! > > try to buy one to trye it out before buying fore hole company.. > > > > /MVH Fille > > > > > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DeadAgi
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Edwards wrote: > On Mon, 24 Dec 2007, Matt Riddell wrote: > >> There is some debate over this but the gist of it is that AGI will send >> a kill to your script if the channel is hung up. DeadAGI won't. > > res_agi.c says SIGHUP, not SIGKILL. Correct. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHb2LODQNt8rg0Kp4RAiIMAJ4u0fl9KfqkL9ViG4RPzjdlNteSbwCeOKmh WD4Ygd1aMNj7Rastk7bU9Po= =595g -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
