Re: [asterisk-users] Skype for Asterisk: Public Beta available
Pascal Bruno wrote:
Unfortunately for me, I cannot register my license. Kept saying:
Could not generate Host-ID.
Make sure that you have eth0 enabled.
Any help would be appreciated
It uses the same licensing scheme as the G.729 licenses (so as soon as
you need to upgrade the machine, or set up LACP or VPN or any other type
of virtual interface or in the case of G.729 you change the codec to a
newer version {since you've upgraded to a new version of asterisk that
doesn't support older ones} that doesn't support the old name for the
codec, you need to re-register).
Or as in your case, it doesn't like the names of the network interfaces.
It's all a total PITA.
Fwiw, the Skype channel driver stopped working on my machine a while
ago. I never did track down the cause.
When res_skypeforasterisk starts, 39 res_skypeforasterisk processes
start and 1 skypewatcher service starts.
If I start it manually after asterisk has started, usually asterisk
segfaults, (not always).
Although Sometimes it starts up properly but can't log anyone in, Either
the user is stated as Logged Out or Connection Error, usually if I type
skype show users I get the following error message:
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20
(Debian 5.0.2 x64 running kernel 2.6.30.2, asterisk 1.6.1.1 and
skypeforasterisk-1.6.1_0.9.10-x86_64)
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Re: [asterisk-users] Skype for Asterisk: Public Beta available
Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
I reported an issue on Mantis (#14). Waiting for an update. http://betareports.digium.com/mantis/view.php?id=14 Emrah Emrah wrote: Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codecs for reading and writing
On 1 Aug 2009, at 22:26, Alex Balashov wrote: Elliot Murdock wrote: Thank you...do you know if IAX can do this? The reason for doing is this is to get over the adsl upload/download discrepancy. While G711 gives terrific quality, it is not always that feasible for the upload direction, which has much more limited bandwidth. Accordingly, it would be possible to use G729 for upload, but keep the higher quality codec, G711, for download. I do not know a great deal about IAX so I will defer to the experts for the definitive word on whether it is possible from the point of view of its formal protocol mechanics. However, poking around the various configuration options for IAX peers on voip-info.org and a few other places suggests that there is no option to do that with IAX, either. It's not really something 99.9% of VoIP users want to do. :-) I think you will find that it may work with Asterisk's IAX implementation. The protocol expects the 2 ends to agree a single symmetrical codec as part of the connection setup, but it doesn't define what actually happens if the codec specified in the first (full frame) voice packet isn't what was agreed. I have a vague memory that if the codec is one that is allowed, asterisk does 'the right thing' issues a warning and uses what it was given. But, as Alex says, there is no clear way to define this in the config files. You would probably do better to use Speex in both directions, but set the encoding quality (in codec.conf ) parameters to be different at the 2 ends. The speex decoder should at the far end should be fine with that. see http://www.voip-info.org/wiki/view/Asterisk+config+codecs.conf Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codecs for reading and writing
Tim Panton wrote: The protocol expects the 2 ends to agree a single symmetrical codec as part of the connection setup, but it doesn't define what actually happens if the codec specified in the first (full frame) voice packet isn't what was agreed. Asterisk only supports symmetric codec configuration on its internal channels, so in Asterisk's IAX2 implementation, if a frame is received from the other endpoint that is not in the 'expected' format a warning is issued and the outbound direction is automatically switched to the same format. The same is done for any protocol using RTP in Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Converting sound files
Hi all, I have a set of sound files that are recorded in 16 Bit 44.1 KHz stereo and I want to convert them into 16 bit 8000 KHz mono so that i can use them in Asterisk. What is the best way of doing that? Many thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Converting sound files
Christian wrote: Hi all, I have a set of sound files that are recorded in 16 Bit 44.1 KHz stereo and I want to convert them into 16 bit 8000 KHz mono so that i can use them in Asterisk. What is the best way of doing that? http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Converting sound files
On linux you can use Sox. Google it and resd the documentation to see how you can convert files from the command line. On windows you can use Switch by NCH Software. Download the trial then you can pay a small fee if you want to keep it. Sent from my iPod On Aug 2, 2009, at 10:30 AM, Christian christia...@runbox.com wrote: Hi all, I have a set of sound files that are recorded in 16 Bit 44.1 KHz stereo and I want to convert them into 16 bit 8000 KHz mono so that i can use them in Asterisk. What is the best way of doing that? Many thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
So what do you think I can do to register my license? I am running
Asterisk 1.6.10 on CentOS 5.
Sent from my iPod
On Aug 2, 2009, at 3:49 AM, Thomas Kenyon dig...@sanguinarius.co.uk
wrote:
Pascal Bruno wrote:
Unfortunately for me, I cannot register my license. Kept saying:
Could not generate Host-ID.
Make sure that you have eth0 enabled.
Any help would be appreciated
It uses the same licensing scheme as the G.729 licenses (so as soon as
you need to upgrade the machine, or set up LACP or VPN or any other
type
of virtual interface or in the case of G.729 you change the codec to a
newer version {since you've upgraded to a new version of asterisk that
doesn't support older ones} that doesn't support the old name for the
codec, you need to re-register).
Or as in your case, it doesn't like the names of the network
interfaces.
It's all a total PITA.
Fwiw, the Skype channel driver stopped working on my machine a while
ago. I never did track down the cause.
When res_skypeforasterisk starts, 39 res_skypeforasterisk processes
start and 1 skypewatcher service starts.
If I start it manually after asterisk has started, usually asterisk
segfaults, (not always).
Although Sometimes it starts up properly but can't log anyone in,
Either
the user is stated as Logged Out or Connection Error, usually if I
type
skype show users I get the following error message:
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20
(Debian 5.0.2 x64 running kernel 2.6.30.2, asterisk 1.6.1.1 and
skypeforasterisk-1.6.1_0.9.10-x86_64)
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Re: [asterisk-users] Skype for Asterisk: Public Beta available
On Sun, Aug 2, 2009 at 8:24 AM, Pascal Brunotipas...@gmail.com wrote: So what do you think I can do to register my license? I am running Asterisk 1.6.10 on CentOS 5. Could not generate Host-ID. Make sure that you have eth0 enabled. The MAC is used in the scheme to register and it looks like it can't be read for some reason. There must be a direct channel to Digium for the support of this kind, though. Have you tried contacting them? [waits for John Todd to chime in here...] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to sniff RTP and SIP traffic only
Wireshark will support this... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Xavier Cardil Sent: Monday, June 29, 2009 5:51 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to sniff RTP and SIP traffic only Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster debugging ? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
On Sun, Aug 2, 2009 at 12:13 PM, randulo spamsucks2...@gmail.com wrote: On Sun, Aug 2, 2009 at 8:24 AM, Pascal Brunotipas...@gmail.com wrote: So what do you think I can do to register my license? I am running Asterisk 1.6.10 on CentOS 5. Could not generate Host-ID. Make sure that you have eth0 enabled. The MAC is used in the scheme to register and it looks like it can't be read for some reason. There must be a direct channel to Digium for the support of this kind, though. Have you tried contacting them? [waits for John Todd to chime in here...] Is eth0 enabled? Is it named eth0? What does ifconfig eth0 tell you? I have seen many Dell servers where the two NICs are labeled eth1 and eth2 or whatever, but in Linux, they are backwards. Eth2 show up as eth0 and eth1 shows up as eth1 in Linux. Wasted a good half hour to forty five minutes trying to figure out why I couldn't get the network up. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Converting sound files
Hi, Many thanks I used it and it worked fine. Christian On 2009-08-02 at 11:21 Pascal Bruno wrote: On linux you can use Sox. Google it and resd the documentation to see how you can convert files from the command line. On windows you can use Switch by NCH Software. Download the trial then you can pay a small fee if you want to keep it. Sent from my iPod On Aug 2, 2009, at 10:30 AM, Christian christia...@runbox.com wrote: Hi all, I have a set of sound files that are recorded in 16 Bit 44.1 KHz stereo and I want to convert them into 16 bit 8000 KHz mono so that i can use them in Asterisk. What is the best way of doing that? Many thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
Well I think thats what the problem was, I dont have it named as eth0. So if your NIC is not labeled eth0 you cannot use skypeforasterisk??? Why cant it just scan you nic handles? Can someone point me to where I can change the NIC name in the source file or something??? On Sun, Aug 2, 2009 at 1:05 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Sun, Aug 2, 2009 at 12:13 PM, randulo spamsucks2...@gmail.com wrote: On Sun, Aug 2, 2009 at 8:24 AM, Pascal Brunotipas...@gmail.com wrote: So what do you think I can do to register my license? I am running Asterisk 1.6.10 on CentOS 5. Could not generate Host-ID. Make sure that you have eth0 enabled. The MAC is used in the scheme to register and it looks like it can't be read for some reason. There must be a direct channel to Digium for the support of this kind, though. Have you tried contacting them? [waits for John Todd to chime in here...] Is eth0 enabled? Is it named eth0? What does ifconfig eth0 tell you? I have seen many Dell servers where the two NICs are labeled eth1 and eth2 or whatever, but in Linux, they are backwards. Eth2 show up as eth0 and eth1 shows up as eth1 in Linux. Wasted a good half hour to forty five minutes trying to figure out why I couldn't get the network up. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
I had that too, I cured it by kill -9 'ing the skypeforasterisk process that was left over from the previous version of the beta. Hope that helps. Tim. On 2 Aug 2009, at 11:20, Emrah wrote: I reported an issue on Mantis (#14). Waiting for an update. http://betareports.digium.com/mantis/view.php?id=14 Emrah Emrah wrote: Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
Hi Tim, I don't have any skypeforasterisk process running. I tried to killall -9 asterisk but it did not solve my issue. Any other suggestions? Thanks for your help, Emrah Tim Panton wrote: I had that too, I cured it by kill -9 'ing the skypeforasterisk process that was left over from the previous version of the beta. Hope that helps. Tim. On 2 Aug 2009, at 11:20, Emrah wrote: I reported an issue on Mantis (#14). Waiting for an update. http://betareports.digium.com/mantis/view.php?id=14 Emrah Emrah wrote: Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Modem
Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modem
Carlos Ruiz Diaz wrote: Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. the digium single fxo cards and clones for about $10 ARE modems. you can get a sip gateway fxo + fxs in one box for about $50 really - how much cheaper do you want ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modem
I did not know that the price was that low. Anyway, for people living really far from USA the price gets incremented twice or more and this is without considering the conversion between currencies. 1 $ = 5100 Gs., not cheap at all. Thanks. On Sun, Aug 2, 2009 at 3:07 PM, jon pounder j...@inline.net wrote: Carlos Ruiz Diaz wrote: Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. the digium single fxo cards and clones for about $10 ARE modems. you can get a sip gateway fxo + fxs in one box for about $50 really - how much cheaper do you want ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modem
Carlos Ruiz Diaz wrote: I did not know that the price was that low. Anyway, for people living really far from USA the price gets incremented twice or more and this is without considering the conversion between currencies. 1 $ = 5100 Gs., not cheap at all. all that stuff is coming from hongkong and china - tons on ebay and auctions in all sorts of currencies with nearly free shipping, so yes it is cheap, does not matter where you live, you just need to look. Thanks. On Sun, Aug 2, 2009 at 3:07 PM, jon pounder j...@inline.net mailto:j...@inline.net wrote: Carlos Ruiz Diaz wrote: Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. the digium single fxo cards and clones for about $10 ARE modems. you can get a sip gateway fxo + fxs in one box for about $50 really - how much cheaper do you want ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T.38 and reinvite
I have a setup with a number of customer Asterisks with T.38 enabled. This works quite well for each customer sending faxes between branch offices. They all have a SIP trunk to a central Asterisk, which connects them to the PSTN through various providers on dedicated lines. I cannot enable reinvite on those SIP trunks, because that would allow calls from the customer's phones to get reinvited and talk directly to the central Asterisk -- and there are firewall rules forbidding that. Is there a way to get ONLY T.38 reinvite without Asterisk trying to get out of the media path? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
I don't know then. My understanding is that the message is caused by the wrong skypeforasterisk process running. - did you (ever) run it as a different user ? If it is a test box, you could try a full reboot. Tim. On 2 Aug 2009, at 19:35, Emrah wrote: Hi Tim, I don't have any skypeforasterisk process running. I tried to killall -9 asterisk but it did not solve my issue. Any other suggestions? Thanks for your help, Emrah Tim Panton wrote: I had that too, I cured it by kill -9 'ing the skypeforasterisk process that was left over from the previous version of the beta. Hope that helps. Tim. On 2 Aug 2009, at 11:20, Emrah wrote: I reported an issue on Mantis (#14). Waiting for an update. http://betareports.digium.com/mantis/view.php?id=14 Emrah Emrah wrote: Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Error
Sorry i think i forgot to mention that i have /var/spool/asterisk as a directory from another server mounted via sshfs. when i don't use a remote directory recording works fine. not sure if this is a permission, but i switched to user asterisk and created new files on the remote directory i can create and overwrite files. any ideas? TIA Regards Ron Ron wrote: Hi All, I'm trying to test asterisk voicemail on recording my own unavailable message, busy message or temporary message. I was looking at the console and saw this message: app_voicemail store_file Memory map failed Then i looked at /var/spool/asterisk/ there were no recorded greetings. what does the error mean? TIA Regards Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modem
On Sun, 2009-08-02 at 14:54 -0400, Carlos Ruiz Diaz wrote: Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. This question has been answered many times, but just for the fun of it I'll answer it again: If PC modems had been ideal telephony cards, we'd still be using them. My own experience with using modems as FXO devices (long before I became a Digium employee) was that they were awful. I encountered problems with echo, half-duplex audio, and lack of far-end disconnect supervision. All of those problems are solved with most modern telelphony cards (except for the ultra-cheap cards, which are still just modems). To put it frankly, I wouldn't wish one of those modems on my worst enemies. Anyway, for people living really far from USA the price gets incremented twice or more and this is without considering the conversion between currencies. 1 $ = 5100 Gs., not cheap at all. I understand that the cards are disproportionately expensive in many parts of the world as compared to the United States, because of the difference in economies. I spent a couple of years in Paraguay in the mid 90s, and know what it's like to pay outrageous prices for specialized electronics just because they have to be imported from other countries. (I'm guessing that you're from Paraguay, based on on the monetary conversion you gave. Does Antelco still dominate the telco market in Paraguay, I wonder?) That being said, the cost per port of the Digium cards (or any of our competitors who design their own cards) is still much lower than what you'd pay for traditional telephony cards, such as those manufactured by Dialogic or Aculab. I know that probably doesn't help you afford to be able to buy a more expensive card, but hopefully you have a better understanding of why we don't use modems as FXO devices. If your time and sanity are worth anything at all, it's a worthwhile investment to buy a good solid telephony card. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to sniff RTP and SIP traffic only
To make your life a little easier, you can use the following filter: sip or sdp or rtp Just insert that into the filter query field in wireshark and it'll show you what you need. On Sun, Aug 2, 2009 at 12:49 PM, Joe Carroll j...@myl2n.com wrote: Wireshark will support this… *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Xavier Cardil *Sent:* Monday, June 29, 2009 5:51 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] how to sniff RTP and SIP traffic only Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster debugging ? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modem
On Sun, Aug 02, 2009 at 03:07:08PM -0400, jon pounder wrote: Carlos Ruiz Diaz wrote: Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. Because nobody bothered writing drivers for any of them. the digium single fxo cards and clones for about $10 ARE modems. you can get a sip gateway fxo + fxs in one box for about $50 At the time they were. Nowadays they cost more. Those specific modems are now obsolete and not manufactured anymore. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.6.7 released
The Astlinux Development Team is happy to announce the release of AstLinux 0.6.7. This release is a security and bugfix release with no new features. All current users of AstLinux are encouraged to upgrade. Current users can upgrade either from the web interface or by issuing the following commands from the CLI. upgrade-run-image check http://mirror.astlinux.org/firmrware (which should report ... Newest available version is: astlinux-0.6.7) then upgrade-run-image upgrade http://mirror.astlinux.org/firmware New users can download the installation images or ISO images from the Sourceforge project site. Note that sourceforge made some changes so not all 0.6.7 versions appear under 'newest files'. You'll need to browse to the bottom of the files page to the 0.6.7 https://sourceforge.net/projects/astlinux/ Release notes are available here: http://www.astlinux.org/releasenotes/0.6.7 Regards, The Astlinux Development Team ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and E1 Cards
Greetings List, i have a new question regarding Asterisk and E1 Cards a client of mine is requiring an Asterisk Server with 2 E1s. the scenario is the following they want 400 extensions to register with the system.. and required 64 concurrent calls. added to it that they are expecting the system to have an IVR to do some DB querying. the setup I have in mind is a Core2duo Server with 3 GB Ram and a Raid0 and a TE220B card. we have not faced this need from a client as we usually provide SIP Services only.. so my questions are the following 1- how many calls my setup will be able to handle? and if it won't handle 2 E1s what is the best server i can get for that? 2- E1 supports Ulaw and Alaw codecs so we won't be needing G729 nor G723 encoding and decoding? or we will have to use such codecs? (I'm concirned about the System resources) Thank you in Advance for your help and support. regards Tarek _ Get free photo software from Windows Live http://www.windowslive.com/online/photos?ocid=PID23393::T:WLMTAGL:ON:WL:en-US:SI_PH_software:082009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and E1 Cards
On Sun, Aug 2, 2009 at 6:37 PM, Tarek Sawah tareksa...@hotmail.com wrote: Greetings List, Greetings i have a new question regarding Asterisk and E1 Cards a client of mine is requiring an Asterisk Server with 2 E1s. the scenario is the following they want 400 extensions to register with the system.. and required 64 concurrent calls. Unless I am mistaking, or you are including internal calls, 2 E1 would handle 62 or 60 if PRI. 400 extensions should be no problem. On the same LAN? added to it that they are expecting the system to have an IVR to do some DB querying. the setup I have in mind is a Core2duo Server with 3 GB Ram and a Raid0 and a TE220B card. Hard to say which would be better, two lower spec (cheaper) boxen setup identically, one as a cold swap. Backup DB, conf, and whatever, nightly. I have done this for many customers. RAID 0 is basically useless for Asterisk and sets yourself up for double chance of disk failure. RAID 1 is the way to go. we have not faced this need from a client as we usually provide SIP Services only.. so my questions are the following 1- how many calls my setup will be able to handle? and if it won't handle 2 E1s what is the best server i can get for that? You can handle that easily unless you are doing heavy codecs like G729 or recording every call. 2- E1 supports Ulaw and Alaw codecs so we won't be needing G729 nor G723 encoding and decoding? or we will have to use such codecs? (I'm concirned about the System resources) You should have said that first ;) A pentium 4 2.8ghz could handle this without breaking a sweat. Thank you in Advance for your help and support. regards Tarek -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and E1 Cards
do you suggest buying a licensed Software from Digium? Date: Sun, 2 Aug 2009 18:53:16 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk and E1 Cards On Sun, Aug 2, 2009 at 6:37 PM, Tarek Sawah tareksa...@hotmail.com wrote: Greetings List, Greetings i have a new question regarding Asterisk and E1 Cards a client of mine is requiring an Asterisk Server with 2 E1s. the scenario is the following they want 400 extensions to register with the system.. and required 64 concurrent calls. Unless I am mistaking, or you are including internal calls, 2 E1 would handle 62 or 60 if PRI. 400 extensions should be no problem. On the same LAN? added to it that they are expecting the system to have an IVR to do some DB querying. the setup I have in mind is a Core2duo Server with 3 GB Ram and a Raid0 and a TE220B card. Hard to say which would be better, two lower spec (cheaper) boxen setup identically, one as a cold swap. Backup DB, conf, and whatever, nightly. I have done this for many customers. RAID 0 is basically useless for Asterisk and sets yourself up for double chance of disk failure. RAID 1 is the way to go. we have not faced this need from a client as we usually provide SIP Services only.. so my questions are the following 1- how many calls my setup will be able to handle? and if it won't handle 2 E1s what is the best server i can get for that? You can handle that easily unless you are doing heavy codecs like G729 or recording every call. 2- E1 supports Ulaw and Alaw codecs so we won't be needing G729 nor G723 encoding and decoding? or we will have to use such codecs? (I'm concirned about the System resources) You should have said that first ;) A pentium 4 2.8ghz could handle this without breaking a sweat. Thank you in Advance for your help and support. regards Tarek -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Get free photo software from Windows Live http://www.windowslive.com/online/photos?ocid=PID23393::T:WLMTAGL:ON:WL:en-US:SI_PH_software:082009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modem
You understand perfectly fine the situation :) . I'm not saying that Paraguay has the worse economy in South-America, but we need to work much harder to get latest technology or to mount a tiny/small laboratory. You will get amized if you see the things that we have done with pieces of hardware considered as garbage in USA :D Does Antelco still dominate the telco market in Paraguay, I wonder. Yes, they changed their name to Copaco for Compania Paraguaya de Comunicaciones. It's basically the same company ruling the whole country. :S Thanks to all for answering my question. On Sun, Aug 2, 2009 at 4:56 PM, Jared Smith jsm...@digium.com wrote: On Sun, 2009-08-02 at 14:54 -0400, Carlos Ruiz Diaz wrote: Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. This question has been answered many times, but just for the fun of it I'll answer it again: If PC modems had been ideal telephony cards, we'd still be using them. My own experience with using modems as FXO devices (long before I became a Digium employee) was that they were awful. I encountered problems with echo, half-duplex audio, and lack of far-end disconnect supervision. All of those problems are solved with most modern telelphony cards (except for the ultra-cheap cards, which are still just modems). To put it frankly, I wouldn't wish one of those modems on my worst enemies. Anyway, for people living really far from USA the price gets incremented twice or more and this is without considering the conversion between currencies. 1 $ = 5100 Gs., not cheap at all. I understand that the cards are disproportionately expensive in many parts of the world as compared to the United States, because of the difference in economies. I spent a couple of years in Paraguay in the mid 90s, and know what it's like to pay outrageous prices for specialized electronics just because they have to be imported from other countries. (I'm guessing that you're from Paraguay, based on on the monetary conversion you gave. Does Antelco still dominate the telco market in Paraguay, I wonder?) That being said, the cost per port of the Digium cards (or any of our competitors who design their own cards) is still much lower than what you'd pay for traditional telephony cards, such as those manufactured by Dialogic or Aculab. I know that probably doesn't help you afford to be able to buy a more expensive card, but hopefully you have a better understanding of why we don't use modems as FXO devices. If your time and sanity are worth anything at all, it's a worthwhile investment to buy a good solid telephony card. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modem
Yes, they changed their name to Copaco for Compania Paraguaya de Comunicaciones. It's basically the same company ruling the whole country. : Oh, like ATT and Verizon here. :-( Please pardon the editorial comment, list. Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 and reinvite
On Sun, Aug 2, 2009 at 3:30 PM, Benny Amorsenbenny+use...@amorsen.dk wrote: Is there a way to get ONLY T.38 reinvite without Asterisk trying to get out of the media path? That's an excellent question. As you've realized, T.38 works by initializing the SIP connection as audio over a chosen codec, and then if T.38 is offered there's a reinvite to initiate the T.38 session. Because of this behavior, and because you want to disable the reinvite for normal audio calls, I think you will have a to use a separate SIP trunk for faxing. I can't think of any other way to do it. So a second parallel SIP trunk for each location sounds like the your way out of this problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.0.11-rc2, 1.6.1.2, 1.6.1.3-rc1, and 1.6.2.0-beta4 Release Announcement
The Asterisk Development Team is pleased to announce the the second release candidate of 1.6.0.11, the release of 1.6.1.2, the first release candidate of 1.6.1.3, and the fourth beta of 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ . The release of 1.6.1.2 fixes a remote crash security vulnerability in the RTP stack. The related security advisory AST-2009-004 has been released along with this announcement. Please read that advisory for more information. The release candidates and betas, in addition to other fixes, contain a major re-work of the T.38 support in Asterisk. If you've been having trouble with T.38 in the 1.6 series, you are strongly encouraged to try one of these release candidates to determine if these changes fixed your T.38 issues. For a full list of changes in these releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.0.11-rc2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.3-rc1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.0-beta4 Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.6 call forwarding
2009/7/31 pepesz76 pepes...@o2.pl
Dear All,
I'n trying to make a simple call forwarding, however I have small
problem when evaluating an expresion.
Here is my extensions.conf
...
; Unconditional Call Forward
exten = _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
exten = _#21*X.,2,Hangup()
exten = #21#,1,Set(ignored=${DB_DELETE(CFIM/${CALLERID(num)})})
exten = #21#,2,Hangup()
...
exten = 50,1,Set(CFIM=${DB(CFIM/${EXTEN})})
exten = 50,n,GotoIf($[${CFIM}=]?start) ;- THIS IS WRONG, but not
sure what should it look like?
exten = 50,n,Dial(SIP/${CFIM},30)
exten = 50,n,Dial(SIP/${EXTEN},30
...
First part properly sets and deletes string in database
Second part works as the forwarding is set, however if it is not set
then CFIM is empty and I got:
WARNING[9752]: ast_expr2.fl:434 ast_yyerror: ast_yyerror(): syntax error:
syntax error, unexpected '=', expecting $end; Input:
=''
Can someone suggest the solution?
What it's doing in this case is first extracting the variable value then
performing the evaluation of the condition, so, when there is no value set,
what it's effectivly trying in the condition is $[=], hence the unexpected
'=' message. If you quote the variable, i.e. $[${CFIM}=] then in the
event the variable is unset or contains an empty value the condition will
still be at least $[=] and it should work...
d
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[asterisk-users] AST-2009-004: Remote Crash Vulnerability in RTP stack
Asterisk Project Security Advisory - AST-2009-004 ++ | Product| Asterisk| |--+-| | Summary| Remote Crash Vulnerability in RTP stack | |--+-| | Nature of Advisory | Exploitable Crash | |--+-| |Susceptibility| Remote unauthenticated sessions | |--+-| | Severity | Critical| |--+-| |Exploits Known| No | |--+-| | Reported On | July 27, 2009 | |--+-| | Reported By | Marcus Hunger hunger AT sipgate DOT de| |--+-| | Posted On | August 2, 2009 | |--+-| | Last Updated On| August 2, 2009 | |--+-| | Advisory Contact | Mark Michelson mmichelson AT digium DOT com | |--+-| | CVE Name | | ++ ++ | Description | An attacker can cause Asterisk to crash remotely by | | | sending malformed RTP text frames. While the attacker| | | can cause Asterisk to crash, he cannot execute arbitrary | | | remote code with this exploit. | ++ ++ | Resolution | Users should upgrade to a version listed in the | || Corrected In section below. | ++ ++ | Affected Versions| || |Product| Release Series | | |---++---| | Asterisk Open Source | 1.2.x | Unaffected| |---++---| | Asterisk Open Source | 1.4.x | Unaffected| |---++---| | Asterisk Open Source | 1.6.x | All 1.6.1 versions| |---++---| |Asterisk Addons| 1.2.x | Unaffected| |---++---| |Asterisk Addons| 1.4.x | Unaffected| |---++---| |Asterisk Addons| 1.6.x | Unaffected| |---++---| | Asterisk Business Edition | A.x.x | Unaffected| |---++---| | Asterisk Business Edition | B.x.x | Unaffected| |---++---| | Asterisk Business Edition | C.x.x | Unaffected| |---++---| | AsteriskNOW | 1.5 | Unaffected| |---++---| | s800i (Asterisk Appliance) | 1.2.x | Unaffected| ++ ++ |
Re: [asterisk-users] Asterisk 1.6.0.11-rc2, 1.6.1.2, 1.6.1.3-rc1, and 1.6.2.0-beta4 Release Announcement
Faxing over SIP never worked for me. The faxes would always fail. When I saw the information about T.38, I decided to immediately upgrade to 1.6.0.11-rc2 from 1.6.0.10 and try it. I was amazed. Without having to change anything in my configuration faxes just worked. I have tested it with multiple faxes, short and long, and faxes with images and they all came through. Well done guys. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Team Sent: Monday, 3 August 2009 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6.0.11-rc2, 1.6.1.2, 1.6.1.3-rc1,and 1.6.2.0-beta4 Release Announcement The Asterisk Development Team is pleased to announce the the second release candidate of 1.6.0.11, the release of 1.6.1.2, the first release candidate of 1.6.1.3, and the fourth beta of 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ . The release of 1.6.1.2 fixes a remote crash security vulnerability in the RTP stack. The related security advisory AST-2009-004 has been released along with this announcement. Please read that advisory for more information. The release candidates and betas, in addition to other fixes, contain a major re-work of the T.38 support in Asterisk. If you've been having trouble with T.38 in the 1.6 series, you are strongly encouraged to try one of these release candidates to determine if these changes fixed your T.38 issues. For a full list of changes in these releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 1.6.0.11-rc2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 1.6.1.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 1.6.1.3-rc1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 1.6.2.0-beta4 Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] User Authentication in sip.conf
Dear all, I want to setup the incoming calls, that don't use authentication in sip.conf file. My configurations as follows, [2000] type=peer host=dynamic insecure=port,invite; (both) context=Testing But when I call '2000', I noticed the following message in Asterisk console, NOTICE[5702]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user Velusamy sip:7...@192.168.1.222sip%3a...@192.168.1.222 ;tag=yj66acQcycvrN What would be the problem?? Please help me to solve this problem. Best Regards, Velusamy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Options Enter Leave Sound
2009/7/24 Stefan Schmidt s...@sil.at Hello, i´ve a question about the Meetme Options. How could i play a enter and leave sound but without recording the user name first. I just want a User joined conferenc and a user leaved. With the i or I Option i have to record the name first. Is there any way of doing this? As i can see in the Meetme help the background agi couldnt be used on non dahdi channel which i will have on this server cause there is no direct Pri link, just SIP. By default join and leave sounds are played when someone joins and leaves the conference... d ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing over Carrier SIP trunk/g711 ?
Jason Aarons (US) wrote: Anyone have a customer sending/receiving multi-page faxes over Verizon Business SIP trunk/g711 ? Verizon Business indicates they don’t support it, and I have 2 recent customers that it doesn’t work for, and 1 current large customer telling me he’s going to make it work grin. The issues is the latency/jitter on fax/g711 over Verizon Business seems to spit out only 11 pages of a 15 page fax. Anyone having faxing over PSTN SIP over G711 that is working? Any advice? Read: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
