Re: [asterisk-users] GTalk Channel
How many ports have you forwarded for the * ? (in rtp.conf) If a limited amount (50-100), try to forward more. 2009/1/29 GNUbie gnu...@gmail.com Hello all, In addition to my previous e-mail, below is a more verbosed messages I got on my Asterisk shell when calling from another GTalk User ID to the Asterisk-1.4.21.2 box: pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=49 from=cal...@gmail.com/Talk.v1041B79926Bsession type=initiate id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;description xml:lang=en xmlns=http://www.google.com/session/phone;payload-type id=103 name=ISAC clockrate=16000/payload-type id=97 name=IPCMWB clockrate=16000 bitrate=8/payload-type id=99 name=speex clockrate=16000 bitrate=22000/payload-type id=4 name=G723 clockrate=8000 bitrate=6300/payload-type id=98 name=speex clockrate=8000 bitrate=11000/payload-type id=100 name=EG711U clockrate=8000 bitrate=64000/payload-type id=101 name=EG711A clockrate=8000 bitrate=64000/payload-type id=0 name=PCMU clockrate=8000 bitrate=64000/payload-type id=8 name=PCMA clockrate=8000 bitrate=64000/payload-type id=13 name=CN clockrate=8000/payload-type id=102 name=iLBC clockrate= JABBER: gtalk INCOMING: 8000 bitrate=13300/payload-type id=106 name=telephone-event clockrate=8000//descriptiontransport xmlns=http://www.google.com/transport/p2p//session/iq [Jan 29 11:18:24] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 11:18:24] WARNING[1303]: chan_gtalk.c:971 gtalk_alloc: Out of RTP sessions? [Jan 29 11:18:24] WARNING[1303]: chan_gtalk.c:1175 gtalk_newcall: Unable to allocate gtalk structure! pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=51 from=cal...@gmail.com/Talk.v1041B79926Bsession type=transport-info id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;transport xmlns=http://www.google.com/transport/p2p;candidate name=rtp address=10.20.1.151 port=1587 preference=1 username=RrBBqm7MeJW2zTgi protocol=udp generation=0 password=OjLNI9dyFLqqBi/Y type=local network=0//transport/session/iq pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=52 from=cal...@gmail.com/Talk.v1041B79926Bsession type=transport-info id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;transport xmlns=http://www.google.com/transport/p2p;candidate name=rtp address=219.74.65.168 port=1588 preference=0.9 username=sHhE4y2GwRBmLQUB protocol=udp generation=0 password=BYAvdVRiU94RVOJW type=stun network=0//transport/session/iq pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=54 from=cal...@gmail.com/Talk.v1041B79926Bsession type=terminate id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session//iq [Jan 29 11:18:40] NOTICE[1303]: chan_gtalk.c:783 gtalk_hangup_farend: Whoa, didn't find call! JABBER: gtalk OUTGOING: iq type='result' from='ast...@gmail.com/asteriskE2D976CC' to='cal...@gmail.com/Talk.v1041B79926B' id='54'/ JABBER: gtalk INCOMING: pbx*CLI JABBER: gtalk INCOMING: presence from=cal...@gmail.com/Talk.v1041B79926B to=ast...@gmail.compriority24/priorityc node=http://www.google.com/xmpp/client/caps; ver=1.0.0.104 ext=share-v1 voice-v1 xmlns=http://jabber.org/protocol/caps/x stamp=20090129T03:17:52 xmlns=jabber:x:delay/status/x xmlns=vcard-temp:x:updatephoto8939f8f8ed0a9cd794e9e3c7065c2cc80fa9dbf0/photo/x/presence pbx*CLI JABBER: gtalk INCOMING: presence from=cal...@gmail.com/Talk.v1041B79926B type=unavailable to=ast...@gmail.com/ pbx*CLI Thank you in advance. Regards, Marvin On Thu, Jan 29, 2009 at 10:47 AM, GNUbie gnu...@gmail.com wrote: Hello all, It used to work on calling my GTalk ID from another GTalk user. But now that I tried calling it again, the caller hears only a ringtone and disconnected after a few rings. The messages on my Asterisk-1.4.21.2 are the following: [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 10:37:51] WARNING[1303]: chan_gtalk.c:971 gtalk_alloc: Out of RTP sessions? [Jan 29 10:37:51] WARNING[1303]: chan_gtalk.c:1175 gtalk_newcall: Unable to allocate gtalk structure! [Jan 29 10:38:06] NOTICE[1303]: chan_gtalk.c:783 gtalk_hangup_farend: Whoa, didn't find call! Any idea? Thank you in advance. Regards, GNUbie ___ -- 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
Re: [asterisk-users] Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
1.4.23.1 doesn't seem to work for me. I did an in-place upgrade of Asterisk 1.4.21 and upgraded to the latest zaptel as well. Incoming calls stopped working. Whenever an extension was trying to pickup the phone by doing a group pickup with *8 the extension just got dead audio and the next phone in the group stared ringing. I was running 1.4.23.1 with the latest FreePBX. 1.4.22.2 is working fine. (Yes i know zaptel was replaced by DAHDI but upgrading is a PITA) On Fri, 23 Jan 2009, Asterisk Development Team wrote: The Asterisk.org development team has announced the release of Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5. These releases are available for immediate download from http://downloads.digium.com/. This update for Asterisk includes a security fix for chan_iax2. Please see the associated security adivisory for more details: http://downloads.digium.com/pub/security/AST-2009-001.html These updates are a fix to a previous security release (released as versions 1.2.31, 1.4.22.1, and 1.6.0.3). The new versions are being released after additional testing revealed some issues with the way that scanning for users was blocked. Those issues have been corrected in this release. This security issue affects the 1.2, 1.4, and 1.6 series of Asterisk. Also note, that Asterisk 1.6.0.4-rc1 was released yesterday prior to the security update. That release has been removed as there will be no 1.6.0.4 release, but rather will be reincarnated as 1.6.0.6-rc1. The reason for the dead release is to avoid 5 digit release numbers. ChangeLogs for the various releases are available at: http://downloads.digium.com/pub/asterisk/ChangeLog-1.2.31.1 http://downloads.digium.com/pub/asterisk/ChangeLog-1.4.22.2 http://downloads.digium.com/pub/asterisk/ChangeLog-1.4.23.1 http://downloads.digium.com/pub/asterisk/ChangeLog-1.6.0.5 Thank you for your continued support of Asterisk! ___ -- 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 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
On Thursday 29 January 2009 09:23:41 Remco Barendse wrote: 1.4.23.1 doesn't seem to work for me. I did an in-place upgrade of Asterisk 1.4.21 and upgraded to the latest zaptel as well. Incoming calls stopped working. Whenever an extension was trying to pickup the phone by doing a group pickup with *8 the extension just got dead audio and the next phone in the group stared ringing. Yeah. Thats http://bugs.digium.com:80/view.php?id=14206 I'm also concerned about that one: http://bugs.digium.com:80/view.php?id=13488 cheers t. ___ -- 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] Dynamically turn on/off echo canceller
Hi guys, Is it possible in * to dynamically turn on/off echo canceller from a dial plan on a specific channel? Something like: exten = s,1,ExecIf(shittyline=true?enableec|disableec) Regards, Alex ___ -- 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] CDR when replaces
Hi all, I have a problem with billing in Asterisk. I am using an Asterisk box as a PSTN gateway with BRI cards with a SIP trunk to an OpenSER proxy. The problem is when I have to transfer an [OpenSER] internal call to PSTN. After transfer, the CDR generated by Asterisk only has the information of the first INVITE and no information about the INVITE with the Replaces is include. In my system, in transfers, billing is charge to the one that enjoys the transfer, so with that CDR I can not follow that policy. Does anyone knows how to get the information in the replaces-INVITE in Asterisk to put into the CDR? Thanks -- Arturo Díaz ___ -- 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] Benqtelecom in cdr log
hi all, I've been noticing some unknown traffic in my cdr logs approximately 50 quantity like this; 2009-01-29 08:17:24 SIP/96.9.1... BenQTelecom BenQ Telecom s ANSWERED 00:20 I wonder that have anyone ever faced like this problem ? I think it's a security problem, when I googled I have found one topic explanation in english http://www.pbxinaflash.com/forum/showthread.php?t=3265 but without solution. Has anyone any recommend to solve it ? Thanks in advance Yavuzhan ___ -- 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] Wanted information
Hi, Ambarish here from India, New (beginner) to asterisk here, Wanted to know how can I install asterisk on Windows XP SP2, with AMD Athlon 64 processor 2.20 Ghz, 512 RAM Can anybody help / guide me in this? Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- 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] Wanted information
On 29 Jan 2009, at 10:47, ambarish.deshm...@wipro.com ambarish.deshm...@wipro.com wrote: Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com Cut this crap from your email. Its wasting my bandwidth. ___ -- 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] CDR when replaces
You should read this. http://lists.digium.com/pipermail/asterisk-users/2008-January/204856.html Regards, Greyman. ___ -- 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] Don't get asterisk to run behind NAT router
Hi people! I am not getting smart getting asterisk 1.6 behind a NAT to run. 1. I enabled IP forwarding on debian linux 2. told asterisk in general that he is behind NAT and mentioned him his external static IP Adress as well his domain in the outside world. If a client who is connected with a DSL modem calls me, a grandstream module in the LAN behind the router, in the same network asterisk is running at, takes the call. but we can't hear / talk with each other. Ay ideas to get this thing solved?! My general section in sip.conf: [general] port=5060 bindaddr=0.0.0.0 localnet=192.168.1.0/255.255.255.0 externip=85.183.112.3 externhost=voipfax.higazi-it.com allowtransfer=yes qualify=yes nat=yes [2006] type=friend secret=frank host=dynamic context=nurintern nat=no [2007] type=friend secret=jochen host=192.168.1.2 context=nurintern nat=yes ___ -- 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] Wanted information
Cut this crap from your email. Its wasting my bandwidth. +1 ___ -- 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] Wanted information
Hyelo Ambarish: http://www.asteriskwin32.com/ Go for it. Kinjal Dixit On Thu, Jan 29, 2009 at 4:17 PM, ambarish.deshm...@wipro.com wrote: Hi, Ambarish here from India, New (beginner) to asterisk here, Wanted to know how can I install asterisk on Windows XP SP2, with AMD Athlon 64 processor 2.20 Ghz, 512 RAM Can anybody help / guide me in this? Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- 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 -- http://www.linkedin.com/in/kinjaldixit open networker ___ -- 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] blind transfer on hook-flash from SIP phone
Kevin P. Fleming wrote: This is because hook-flash support is in chan_zap, not in the core of Asterisk. There is no need to support hook-flash on any other channel type, because every other channel type supported by Asterisk has its own (more reliable) methods of signaling. I'm seeing many discussions on how to provision Blind Transfer, Attendant Transfer and Call Parking services using keypad keys # and * to invoke the services, however its not clear to me how can it be possible without interfering to normal call when service is not being invoked. For instance, I usually access an external IVR system that uses # and * signals in its menu for interaction with the caller. Let's say I provision a SIP phone (that do not support services like Transfer locally) to have the services provided by Asterisk. May question is, If I call to this IVR system and signals # and/or * are required for interaction with the system, what will happen when I dial # or *? Is there any alternative to invoke mid-call services without using the # and * signals? I was expecting to use Hook-Flash either via INFO or RTP telephone-event. Thanks a lot! Marcelo. ___ -- 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] Wanted information
i hope it isnt for a production server. David 2009/1/29 ambarish.deshm...@wipro.com Hi, Ambarish here from India, New (beginner) to asterisk here, Wanted to know how can I install asterisk on Windows XP SP2, with AMD Athlon 64 processor 2.20 Ghz, 512 RAM Can anybody help / guide me in this? Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Wanted information
Ambarish, no you cannot install it on a PC running windows XP, it' needs it's own dedicated server. You can run it from a virtual machine for testing/learning purposes but for real world implementation it will need it's own computer. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of ambarish.deshm...@wipro.com Sent: Thursday, 29 January 2009 5:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Wanted information Hi, Ambarish here from India, New (beginner) to asterisk here, Wanted to know how can I install asterisk on Windows XP SP2, with AMD Athlon 64 processor 2.20 Ghz, 512 RAM Can anybody help / guide me in this? Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- 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 1.6 dahdi only?
Wilton Helm wrote: I still am not quite on the same page with you, though. There are a lot of commands that aren't function calls that go into various config files. The most basic and obvious one is exten There must be a hundred of these and I don't know where they are listed with all acceptable parameters and ranges and what they do and why. There are examples to get one started, but I don't think I can put my hands on even a definitive definition of exten. Am I making any sense? Maybe these are called variables or something. Are you talking about the 'exten' in this example? exten = s,1,NoOp() If so, then I'd encourage you to read the first few pages of Dialplan Basics a couple more times (Chapter 5). In the PDF, it is listed as page 119, and to answer your question specifically, check out page 122 near the top: The syntax for an extension is the word exten, followed by an arrow formed by the equals sign and the greater-than sign, like this: exten = This is followed by the name (or number) of the extension. When dealing with traditional telephone systems, we tend to think of extensions as the numbers you would dial to make another phone ring. In Asterisk, you get a whole lot more; for example, extension names can be any combination of numbers and letters. Over the course of this chapter and the next, we’ll use both numeric and alphanumeric extensions. Hope that helps, Leif Madsen. ___ -- 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] Don't get asterisk to run behind NAT router
Hi Tamer Try to configure canreinvite=no or canreinvite=nonat. Regards Dimitar Tamer Higazi написа: Hi people! I am not getting smart getting asterisk 1.6 behind a NAT to run. 1. I enabled IP forwarding on debian linux 2. told asterisk in general that he is behind NAT and mentioned him his external static IP Adress as well his domain in the outside world. If a client who is connected with a DSL modem calls me, a grandstream module in the LAN behind the router, in the same network asterisk is running at, takes the call. but we can't hear / talk with each other. Ay ideas to get this thing solved?! My general section in sip.conf: [general] port=5060 bindaddr=0.0.0.0 localnet=192.168.1.0/255.255.255.0 externip=85.183.112.3 externhost=voipfax.higazi-it.com allowtransfer=yes qualify=yes nat=yes [2006] type=friend secret=frank host=dynamic context=nurintern nat=no [2007] type=friend secret=jochen host=192.168.1.2 context=nurintern nat=yes ___ -- 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 smime.p7s Description: S/MIME Cryptographic 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] GTalk Channel
Hello Grygoriy, I am forwarding UDP ports from 1 to 10100. That only means that I am forwarding 101 ports. Please take note also that when I tried calling the GTalk ID, the Asterisk box was idle or there was no any other on-going calls. Regards, GNUbie On Thu, Jan 29, 2009 at 4:15 PM, Grygoriy Dobrovolskyy megaho...@gmail.com wrote: How many ports have you forwarded for the * ? (in rtp.conf) If a limited amount (50-100), try to forward more. 2009/1/29 GNUbie gnu...@gmail.com Hello all, In addition to my previous e-mail, below is a more verbosed messages I got on my Asterisk shell when calling from another GTalk User ID to the Asterisk-1.4.21.2 box: pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=49 from=cal...@gmail.com/Talk.v1041B79926Bsession type=initiate id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;description xml:lang=en xmlns=http://www.google.com/session/phone;payload-type id=103 name=ISAC clockrate=16000/payload-type id=97 name=IPCMWB clockrate=16000 bitrate=8/payload-type id=99 name=speex clockrate=16000 bitrate=22000/payload-type id=4 name=G723 clockrate=8000 bitrate=6300/payload-type id=98 name=speex clockrate=8000 bitrate=11000/payload-type id=100 name=EG711U clockrate=8000 bitrate=64000/payload-type id=101 name=EG711A clockrate=8000 bitrate=64000/payload-type id=0 name=PCMU clockrate=8000 bitrate=64000/payload-type id=8 name=PCMA clockrate=8000 bitrate=64000/payload-type id=13 name=CN clockrate=8000/payload-type id=102 name=iLBC clockrate= JABBER: gtalk INCOMING: 8000 bitrate=13300/payload-type id=106 name=telephone-event clockrate=8000//descriptiontransport xmlns=http://www.google.com/transport/p2p//session/iq [Jan 29 11:18:24] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 11:18:24] WARNING[1303]: chan_gtalk.c:971 gtalk_alloc: Out of RTP sessions? [Jan 29 11:18:24] WARNING[1303]: chan_gtalk.c:1175 gtalk_newcall: Unable to allocate gtalk structure! pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=51 from=cal...@gmail.com/Talk.v1041B79926Bsession type=transport-info id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;transport xmlns=http://www.google.com/transport/p2p;candidate name=rtp address=10.20.1.151 port=1587 preference=1 username=RrBBqm7MeJW2zTgi protocol=udp generation=0 password=OjLNI9dyFLqqBi/Y type=local network=0//transport/session/iq pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=52 from=cal...@gmail.com/Talk.v1041B79926Bsession type=transport-info id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;transport xmlns=http://www.google.com/transport/p2p;candidate name=rtp address=219.74.65.168 port=1588 preference=0.9 username=sHhE4y2GwRBmLQUB protocol=udp generation=0 password=BYAvdVRiU94RVOJW type=stun network=0//transport/session/iq pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=54 from=cal...@gmail.com/Talk.v1041B79926Bsession type=terminate id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session//iq [Jan 29 11:18:40] NOTICE[1303]: chan_gtalk.c:783 gtalk_hangup_farend: Whoa, didn't find call! JABBER: gtalk OUTGOING: iq type='result' from='ast...@gmail.com/asteriskE2D976CC' to='cal...@gmail.com/Talk.v1041B79926B' id='54'/ JABBER: gtalk INCOMING: pbx*CLI JABBER: gtalk INCOMING: presence from=cal...@gmail.com/Talk.v1041B79926B to=ast...@gmail.compriority24/priorityc node=http://www.google.com/xmpp/client/caps; ver=1.0.0.104 ext=share-v1 voice-v1 xmlns=http://jabber.org/protocol/caps/x stamp=20090129T03:17:52 xmlns=jabber:x:delay/status/x xmlns=vcard-temp:x:updatephoto8939f8f8ed0a9cd794e9e3c7065c2cc80fa9dbf0/photo/x/presence pbx*CLI JABBER: gtalk INCOMING: presence from=cal...@gmail.com/Talk.v1041B79926B type=unavailable to=ast...@gmail.com/ pbx*CLI Thank you in advance. Regards, Marvin On Thu, Jan 29, 2009 at 10:47 AM, GNUbie gnu...@gmail.com wrote: Hello all, It used to work on calling my GTalk ID from another GTalk user. But now that I tried calling it again, the caller hears only a ringtone and disconnected after a few rings. The messages on my Asterisk-1.4.21.2 are the following: [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 10:37:51] WARNING[1303]: chan_gtalk.c:971 gtalk_alloc: Out of RTP sessions? [Jan 29 10:37:51] WARNING[1303]: chan_gtalk.c:1175 gtalk_newcall: Unable to allocate gtalk structure! [Jan 29 10:38:06] NOTICE[1303]: chan_gtalk.c:783 gtalk_hangup_farend: Whoa, didn't find call! Any idea? Thank you in advance. Regards, GNUbie
Re: [asterisk-users] Dropping incompatible voice frame
Thanks, placing: Disallow=all Allow=ulaw In the specific iaxy device context fixed it. I had always thought that allowing all possible valid codecs under the general context would work and the devices would sort it out upon handshake. Guess not. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven J. Douglas Sent: Wednesday, January 28, 2009 9:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dropping incompatible voice frame Don't use g729 in the iax.conf for the IAXY device. It doesn't support it. Regards, Steve Adam Robins wrote: I am using a Polycom SIP phone (ext 2042) to call an analog phone connected via an IAXY (ext 2120). The analog phone rings, and when I answer, I can hear the person speaking on the SIP phone, but they cannot hear me. However, if I originate the call from the analog phone to the SIP phone, it works just fine. In SIP.conf: Disallow=all Allow=g729 Allow=ulaw Canreinvite=no In IAX.conf: Disallow=all Allow=ulaw Allow=g729 Transfer=no Codecpriority=host CLI shows: [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Executing [2...@international:1] Dial(SIP/2042-b7b0cc88, IAX2/2120|12|oWwtT) in new stack [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Called 2120 [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Call accepted by 192.168.2.61 (format ulaw) [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Format for call is ulaw [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- IAX2/2120-3849 is ringing [Jan 28 16:04:33] VERBOSE[3428] logger.c: [Jan 28 16:04:33] -- IAX2/2120-3849 answered SIP/2042-b7b0cc88 [Jan 28 16:04:33] NOTICE[3428] channel.c: Dropping incompatible voice frame on IAX2/2120-3849 of format g729 since our native format has changed to 0x4 (ulaw) [Jan 28 16:04:41] VERBOSE[3428] logger.c: [Jan 28 16:04:41] -- Hungup 'IAX2/2120-3849' This is Asterisk 1.4.22, but it also happened on 1.2.4. If I call an IAX2/ulaw softphone from the SIP phone, it works fine. Could it be something in the IAXY provisioning? Any ideas are appreciated. 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 ___ -- 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] Wanted information
I suggest checking out FreeSwitch, it is truly platform independant. While not designed to be a PBX, it is gaining momentum everyday, has many pros and a few cons over Asterisk and runs on Windows too (your goal). I think FS, given time will surpass Asterisk in more areas than simply technical (SIP setups and tear downs rapidly, more simultaneous calls) Another plus is that some companies are strictly M$ shops and will not allow (through corp policy) *nix boxen since it requires a different skillset than an off the shelf, dime a dozen MCP or MCSE. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Thu, Jan 29, 2009 at 7:48 AM, Dean Collins d...@cognation.net wrote: Ambarish, no you cannot install it on a PC running windows XP, it' needs it's own dedicated server. You can run it from a virtual machine for testing/learning purposes but for real world implementation it will need it's own computer. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of ambarish.deshm...@wipro.com Sent: Thursday, 29 January 2009 5:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Wanted information Hi, Ambarish here from India, New (beginner) to asterisk here, Wanted to know how can I install asterisk on Windows XP SP2, with AMD Athlon 64 processor 2.20 Ghz, 512 RAM Can anybody help / guide me in this? Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- 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] Wanted information
Sorry, I agree with Dean Collins and David BUT: is it very friendly to said 'Cut this etc etc' - these comments also use my bandwith. i'm loosing some time reading theese posts +1 +2. why not + 99 ? one shot and be quiet... thank's a lot (hum. me too, i apologize, but too much comments on the list..) - Original Message - From: Dean Collins d...@cognation.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 29, 2009 1:48 PM Subject: Re: [asterisk-users] Wanted information Ambarish, no you cannot install it on a PC running windows XP, it' needs it's own dedicated server. You can run it from a virtual machine for testing/learning purposes but for real world implementation it will need it's own computer. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of ambarish.deshm...@wipro.com Sent: Thursday, 29 January 2009 5:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Wanted information Hi, Ambarish here from India, New (beginner) to asterisk here, Wanted to know how can I install asterisk on Windows XP SP2, with AMD Athlon 64 processor 2.20 Ghz, 512 RAM Can anybody help / guide me in this? Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- 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] GTalk Channel
And what ports have you set in rtp.conf ? i suppose they are the same ? Try to search if there are no spercial jabber ports to open. 2009/1/29 GNUbie gnu...@gmail.com Hello Grygoriy, I am forwarding UDP ports from 1 to 10100. That only means that I am forwarding 101 ports. Please take note also that when I tried calling the GTalk ID, the Asterisk box was idle or there was no any other on-going calls. Regards, GNUbie On Thu, Jan 29, 2009 at 4:15 PM, Grygoriy Dobrovolskyy megaho...@gmail.com wrote: How many ports have you forwarded for the * ? (in rtp.conf) If a limited amount (50-100), try to forward more. 2009/1/29 GNUbie gnu...@gmail.com Hello all, In addition to my previous e-mail, below is a more verbosed messages I got on my Asterisk shell when calling from another GTalk User ID to the Asterisk-1.4.21.2 box: pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=49 from=cal...@gmail.com/Talk.v1041B79926Bsession type=initiate id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;description xml:lang=en xmlns=http://www.google.com/session/phone;payload-type id=103 name=ISAC clockrate=16000/payload-type id=97 name=IPCMWB clockrate=16000 bitrate=8/payload-type id=99 name=speex clockrate=16000 bitrate=22000/payload-type id=4 name=G723 clockrate=8000 bitrate=6300/payload-type id=98 name=speex clockrate=8000 bitrate=11000/payload-type id=100 name=EG711U clockrate=8000 bitrate=64000/payload-type id=101 name=EG711A clockrate=8000 bitrate=64000/payload-type id=0 name=PCMU clockrate=8000 bitrate=64000/payload-type id=8 name=PCMA clockrate=8000 bitrate=64000/payload-type id=13 name=CN clockrate=8000/payload-type id=102 name=iLBC clockrate= JABBER: gtalk INCOMING: 8000 bitrate=13300/payload-type id=106 name=telephone-event clockrate=8000//descriptiontransport xmlns=http://www.google.com/transport/p2p//session/iq [Jan 29 11:18:24] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 11:18:24] WARNING[1303]: chan_gtalk.c:971 gtalk_alloc: Out of RTP sessions? [Jan 29 11:18:24] WARNING[1303]: chan_gtalk.c:1175 gtalk_newcall: Unable to allocate gtalk structure! pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=51 from=cal...@gmail.com/Talk.v1041B79926Bsession type=transport-info id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;transport xmlns=http://www.google.com/transport/p2p;candidate name=rtp address=10.20.1.151 port=1587 preference=1 username=RrBBqm7MeJW2zTgi protocol=udp generation=0 password=OjLNI9dyFLqqBi/Y type=local network=0//transport/session/iq pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=52 from=cal...@gmail.com/Talk.v1041B79926Bsession type=transport-info id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;transport xmlns=http://www.google.com/transport/p2p;candidate name=rtp address=219.74.65.168 port=1588 preference=0.9 username=sHhE4y2GwRBmLQUB protocol=udp generation=0 password=BYAvdVRiU94RVOJW type=stun network=0//transport/session/iq pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=54 from=cal...@gmail.com/Talk.v1041B79926Bsession type=terminate id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session//iq [Jan 29 11:18:40] NOTICE[1303]: chan_gtalk.c:783 gtalk_hangup_farend: Whoa, didn't find call! JABBER: gtalk OUTGOING: iq type='result' from='ast...@gmail.com/asteriskE2D976CC' to='cal...@gmail.com/Talk.v1041B79926B' id='54'/ JABBER: gtalk INCOMING: pbx*CLI JABBER: gtalk INCOMING: presence from=cal...@gmail.com/Talk.v1041B79926B to=ast...@gmail.compriority24/priorityc node=http://www.google.com/xmpp/client/caps; ver=1.0.0.104 ext=share-v1 voice-v1 xmlns=http://jabber.org/protocol/caps/x stamp=20090129T03:17:52 xmlns=jabber:x:delay/status/x xmlns=vcard-temp:x:updatephoto8939f8f8ed0a9cd794e9e3c7065c2cc80fa9dbf0/photo/x/presence pbx*CLI JABBER: gtalk INCOMING: presence from=cal...@gmail.com/Talk.v1041B79926B type=unavailable to=ast...@gmail.com/ pbx*CLI Thank you in advance. Regards, Marvin On Thu, Jan 29, 2009 at 10:47 AM, GNUbie gnu...@gmail.com wrote: Hello all, It used to work on calling my GTalk ID from another GTalk user. But now that I tried calling it again, the caller hears only a ringtone and disconnected after a few rings. The messages on my Asterisk-1.4.21.2 are the following: [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 10:37:51] WARNING[1303]: chan_gtalk.c:971 gtalk_alloc: Out of RTP sessions?
Re: [asterisk-users] Don't get asterisk to run behind NAT router
You enabled port forwarding, but have you actually forwarded any ports ? Defaults are tcp 5060 udp 1-2 2009/1/29 Tamer Higazi th9...@googlemail.com Hi people! I am not getting smart getting asterisk 1.6 behind a NAT to run. 1. I enabled IP forwarding on debian linux 2. told asterisk in general that he is behind NAT and mentioned him his external static IP Adress as well his domain in the outside world. If a client who is connected with a DSL modem calls me, a grandstream module in the LAN behind the router, in the same network asterisk is running at, takes the call. but we can't hear / talk with each other. Ay ideas to get this thing solved?! My general section in sip.conf: [general] port=5060 bindaddr=0.0.0.0 localnet=192.168.1.0/255.255.255.0 externip=85.183.112.3 externhost=voipfax.higazi-it.comhttp://192.168.1.0/255.255.255.0externip=85.183.112.3externhost=voipfax.higazi-it.com allowtransfer=yes qualify=yes nat=yes [2006] type=friend secret=frank host=dynamic context=nurintern nat=no [2007] type=friend secret=jochen host=192.168.1.2 context=nurintern nat=yes ___ -- 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] blind transfer on hook-flash from SIP phone
Hi Marcelo - Is there any alternative to invoke mid-call services without using the # and * signals? I was expecting to use Hook-Flash either via INFO or RTP telephone-event. You can change the keys used to invoke a service in features.conf. I know many people use ## or #1 for blind transfer so as to avoid issues with IVR systems. You may also want to change who has access to the various features (caller vs callee). You can do this with flags in your Dial statement. For example, if you have set the 'T' flag, the caller can do a blind transfer. If you only have the 't' flag set (notice lower case) only the person receiving the call can do a blind transfer. - Noah ___ -- 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] RTP/NAT Traffic to private IP
Hi all, I'd like to connect a softphone at home (nat, dynamic-ip) to a sip-phone in the office via asterisk 1.4.21 (nat, fixed-ip). SIP works well, the phone is ringing, but when I pickup the call, there's no audio on both sides. I debugged the rtp-traffic at home. As long as the phone is ringing, everything is fine. But after the pickup, asterisk sends a SIP/SDP package with its private address (192.168.100.10). After the softphone received this package, it tries to send RTP data to this address! Obviously those packages never reach asterisk... Does 'externip' just works for SIP and not for RTP? Where does the the internal IP-address come from and how can I set the right one? My configuration: [general] externip = 85.XXX.XXX.XXX nat = yes localnet = 192.168.100.0/24 [42] deny=0.0.0.0/0.0.0.0 disallow=all type=friend secret=XXX qualify=yes port=5060 pickupgroup= permit=0.0.0.0/0.0.0.0 nat=yes mailbox...@device host=dynamic dtmfmode=rfc2833 dial=SIP/42 context=from-internal canreinvite=no callgroup= callerid=device 42 allow=alaw accountcode= call-limit=50 Regards Holger ___ -- 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] RTP/NAT Traffic to private IP
I have no problem doing that by adding the information you have under [general] to /etc/asterisk/sip_nat.conf On Thu, Jan 29, 2009 at 7:30 AM, Holger Latz t...@globalview.de wrote: Hi all, I'd like to connect a softphone at home (nat, dynamic-ip) to a sip-phone in the office via asterisk 1.4.21 (nat, fixed-ip). SIP works well, the phone is ringing, but when I pickup the call, there's no audio on both sides. I debugged the rtp-traffic at home. As long as the phone is ringing, everything is fine. But after the pickup, asterisk sends a SIP/SDP package with its private address (192.168.100.10). After the softphone received this package, it tries to send RTP data to this address! Obviously those packages never reach asterisk... Does 'externip' just works for SIP and not for RTP? Where does the the internal IP-address come from and how can I set the right one? My configuration: [general] externip = 85.XXX.XXX.XXX nat = yes localnet = 192.168.100.0/24 [42] deny=0.0.0.0/0.0.0.0 disallow=all type=friend secret=XXX qualify=yes port=5060http://0.0.0.0/0.0.0.0disallow=alltype=friendsecret=XXXqualify=yesport=5060 pickupgroup= permit=0.0.0.0/0.0.0.0 nat=yes mailbox...@device host=dynamic dtmfmode=rfc2833 dial=SIP/42 context=from-internal canreinvite=nohttp://0.0.0.0/0.0.0.0nat=yesmailbox...@devicehost=dynamicdtmfmode=rfc2833dial=sip/42context=from-internalcanreinvite=no callgroup= callerid=device 42 allow=alaw accountcode= call-limit=50 Regards Holger ___ -- 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] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
Hi, I am trying to register an asterisk (Asterisk 1) against another one (Asterisk 2). My problem is that the REGISTER message goes without credentials and the Asterisk 2 send a 401 message to the Asterisk 1. How can I configure Asterisk 1 to force it to send credentials? I have tried setting Asterisk 2's IP in the realm field of Asterisk's 1 sip.conf, but it doesn`t work. Any ideas? Thanks Regards ___ -- 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] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
Inter-* registry is done with iax.conf, not sip.conf. sip is for phones/sip-lines. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Imanol Pardavila Sent: Thursday, January 29, 2009 10:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk Hi, I am trying to register an asterisk (Asterisk 1) against another one (Asterisk 2). My problem is that the REGISTER message goes without credentials and the Asterisk 2 send a 401 message to the Asterisk 1. How can I configure Asterisk 1 to force it to send credentials? I have tried setting Asterisk 2's IP in the realm field of Asterisk's 1 sip.conf, but it doesn`t work. Any ideas? Thanks Regards ___ -- 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] Managing codecs
Hi, I`d like to know the best way to manage codecs to let the CPU breath as much as possible. I understand transcoding is a big part of CPU usage on Asterisk, and I have the following situation: - A SIP provider that offers me G729 and ULAW (my choice, both as allowed) - Some of my calls are from G729 enabled phones to outside lines, and I`d like to send those calls using G279 to my provider - Some of my calls are from PSTN (ulaw) and send out to PSTN (to a cell phone let's say, in a find-me-follow-me fashion). I`d like those calls to be sent out in ulaw. Can this be done, or do I have to choose one or the other for all calls? I know I can do that using sip.conf, but I`d like to choose in my dialplan what codec to use according to various rules. Mike ___ -- 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] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
I want to establish a trunk SIP between Asterisk 1 and Asterisk 2, using a sip account (Asterisk 1 acting as a conventional sip user). Thanks Regards Danny Nicholas escribió: Inter-* registry is done with iax.conf, not sip.conf. sip is for phones/sip-lines. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Imanol Pardavila Sent: Thursday, January 29, 2009 10:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk Hi, I am trying to register an asterisk (Asterisk 1) against another one (Asterisk 2). My problem is that the REGISTER message goes without credentials and the Asterisk 2 send a 401 message to the Asterisk 1. How can I configure Asterisk 1 to force it to send credentials? I have tried setting Asterisk 2's IP in the realm field of Asterisk's 1 sip.conf, but it doesn`t work. Any ideas? Thanks Regards ___ -- 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] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
If you are connecting 2 Asterisk boxes and you are setting them both up then use IAX (Intra Asterisk Exchange) instead of SIP. IAX does not pass off the packets to RTP and should fix some of the firewall problems people get. As far as credential passing I am unsure if it will help that. So like mentioned below set up a connection in iax.conf on both ends. There is a chapter in the O'Reily Asterisk the future of telephony book that talks you through an Asterisk to Asterisk connection using IAX. Is there a specific reason you want to use SIP/RTP? Brent Imanol Pardavila wrote: I want to establish a trunk SIP between Asterisk 1 and Asterisk 2, using a sip account (Asterisk 1 acting as a conventional sip user). Thanks Regards Danny Nicholas escribió: Inter-* registry is done with iax.conf, not sip.conf. sip is for phones/sip-lines. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Imanol Pardavila Sent: Thursday, January 29, 2009 10:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk Hi, I am trying to register an asterisk (Asterisk 1) against another one (Asterisk 2). My problem is that the REGISTER message goes without credentials and the Asterisk 2 send a 401 message to the Asterisk 1. How can I configure Asterisk 1 to force it to send credentials? I have tried setting Asterisk 2's IP in the realm field of Asterisk's 1 sip.conf, but it doesn`t work. Any ideas? Thanks Regards ___ -- 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 -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- 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] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
On Thu, 2009-01-29 at 10:15 -0600, Danny Nicholas wrote: Inter-* registry is done with iax.conf, not sip.conf. sip is for phones/sip-lines. That's not quite accurate. Inter-asterisk registrations *can* be done with either the IAX2 or the SIP protocols. -- Jared Smith Digium, Inc. | Training Manager ___ -- 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] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
Paste your register lines (hide pass) 2009/1/29 Imanol Pardavila imanol.pardav...@ibercom.com I want to establish a trunk SIP between Asterisk 1 and Asterisk 2, using a sip account (Asterisk 1 acting as a conventional sip user). Thanks Regards Danny Nicholas escribió: Inter-* registry is done with iax.conf, not sip.conf. sip is for phones/sip-lines. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Imanol Pardavila Sent: Thursday, January 29, 2009 10:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk Hi, I am trying to register an asterisk (Asterisk 1) against another one (Asterisk 2). My problem is that the REGISTER message goes without credentials and the Asterisk 2 send a 401 message to the Asterisk 1. How can I configure Asterisk 1 to force it to send credentials? I have tried setting Asterisk 2's IP in the realm field of Asterisk's 1 sip.conf, but it doesn`t work. Any ideas? Thanks Regards ___ -- 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
[asterisk-users] Eyebeam or Xlite
Lets presume that my both software are open. Xlute and Eyebeam But I want my calls from Asterisk to land only on Eyebeam and Not on xlite. How to set it ? ___ -- 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 Recording Alias
Where did I make mistake ? On Thu, Jan 29, 2009 at 1:07 AM, David @ULC ucoms2...@gmail.com wrote: http://download.vicidial-group.com/svn/agc_2-X/trunk/docs/SCRATCH_INSTALL.txt vi /usr/local/apache2/conf/httpd.conf add the following lines: AddType application/x-httpd-php .php .phtml LoadModule php4_module libexec/libphp5.so or LoadModule php4_module modules/libphp5.so modify the index.html line and add index.php to the list to disable logging, change: CustomLog logs/access_log common to this: CustomLog /dev/null common to enable web browsing of Recordings on Asterisk server, add this: Alias /RECORDINGS/ /var/spool/asterisk/monitorDONE/ Directory /var/spool/asterisk/monitorDONE Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all files *.mp3 Forcetype application/forcedownload /files /Directory - /usr/local/apache2/bin/apachectl start - go to http://your-new-asterisk-server-ipaddress/ to see if it worked - you are done NOTE: If using PHP5 you may need to add the following line to php.ini: short_open_tag = On On Thu, Jan 29, 2009 at 12:13 AM, David @ULC ucoms2...@gmail.com wrote: Modified httf.conf file and added : -- Alias /recordings/ /var/spool/asterisk/monitorDONE/ Directory /var/spool/asterisk/monitorDONE Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all /Directory Created a folder under vicidial as recordings. FULL_RECORDING is also enabled. But I don't see recordings under recording folder. Any guidance ? ___ -- 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] I need help
his problem is that he needs help. On 1/26/09, Jose P. Espinal j...@slackware-es.com wrote: And your problem is... ? Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com mailto:bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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] Eyebeam or Xlite
David @ULC wrote: Lets presume that my both software are open. Xlute and Eyebeam But I want my calls from Asterisk to land only on Eyebeam and Not on xlite. How to set it ? Give each their own SIP credentials. Then in Extensions.conf, when dialing into your extension, send the call to both SIP devices. Then both will ring on your computer and you can decide which to answer. Caution, I have not tested this scenerio, but it should work as long as the two applications are not trying to use the same orginating port numbers to contact your * server. Lyle ___ -- 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] Eyebeam or Xlite
Since you have to register both, just take xlite out of ring groups. Then xlite can dial out, but can't get a call. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC Sent: Thursday, January 29, 2009 11:23 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Eyebeam or Xlite Lets presume that my both software are open. Xlute and Eyebeam But I want my calls from Asterisk to land only on Eyebeam and Not on xlite. How to set it ? ___ -- 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] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
Hi, The SIP messages flow is this: ### AAA.BBB.CCC.DDD: Asterisk 1 IP address EEE.FFF.GGG.HHH: Asterisk 2 IP address ### REGISTER sip:ast2.domain.comSIP/2.0 Via: SIP/2.0/UDP AAA.BBB.CCC.DDD:19646;branch=z9hG4bK0939cc12;rport From: sip:0...@ast2.domain.com;tag=as715628d7 To: sip:0...@ast2.domain.com Call-ID: 13106f3936f01d1c103d5f5230278...@aaa.bbb.ccc.ddd CSeq: 133 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:s...@aaa.bbb.ccc.ddd:19646 Event: registration Content-Length: 0 Using latest REGISTER request as basis request Sending to AAA.BBB.CCC.DDD : 19646 (NAT) Transmitting (NAT) to AAA.BBB.CCC.DDD:19646: SIP/2.0 100 Trying Via: SIP/2.0/UDP AAA.BBB.CCC.DDD:19646;branch=z9hG4bK0939cc12;received=AAA.BBB.CCC.DDD;rport=19646 From: sip:0...@ast2.domain.com;tag=as715628d7 To: sip:0...@ast2.domain.com Call-ID: 13106f3936f01d1c103d5f5230278...@aaa.bbb.ccc.ddd CSeq: 133 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:0...@eee.fff.ggg.hhh Content-Length: 0 Transmitting (NAT) to AAA.BBB.CCC.DDD:19646: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP AAA.BBB.CCC.DDD:19646;branch=z9hG4bK0939cc12;received=AAA.BBB.CCC.DDD;rport=19646 From: sip:0...@ast2.domain.com;tag=as715628d7 To: sip:0...@ast2.domain.com;tag=as5ccb43ac Call-ID: 13106f3936f01d1c103d5f5230278...@aaa.bbb.ccc.ddd CSeq: 133 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7294c1d1 ontent-Length: 0D Asterisk 1 sends an REGISTER without credentials, and Asterisk 2 replies with a 401 message (with Digest algorithm, realm and nonce). I want to configure the Asterisk 1 in order to send REGISTER with credentials. Thanks Regards Grygoriy Dobrovolskyy escribió: Paste your register lines (hide pass) 2009/1/29 Imanol Pardavila imanol.pardav...@ibercom.com mailto:imanol.pardav...@ibercom.com I want to establish a trunk SIP between Asterisk 1 and Asterisk 2, using a sip account (Asterisk 1 acting as a conventional sip user). Thanks Regards Danny Nicholas escribió: Inter-* registry is done with iax.conf, not sip.conf. sip is for phones/sip-lines. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Imanol Pardavila Sent: Thursday, January 29, 2009 10:01 AM To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk Hi, I am trying to register an asterisk (Asterisk 1) against another one (Asterisk 2). My problem is that the REGISTER message goes without credentials and the Asterisk 2 send a 401 message to the Asterisk 1. How can I configure Asterisk 1 to force it to send credentials? I have tried setting Asterisk 2's IP in the realm field of Asterisk's 1 sip.conf, but it doesn`t work. Any ideas? Thanks Regards ___ -- 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 ___ -- 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] early dial: asterisk and ATA
Hi,
I have a set of Grandstream GXW4008 (units of 8 FXS ATAs) and another set of
Linksys SPA8000 (8 FXS ATAs).
The GXW4008 has a neat feature called early dial which allows me to define
a dial pattern as generic as {*X+,#+,X+} (or something similar; the idea is
to match all digits) and send those digits immediately as they are
pressed to Asterisk. As soon as * finds a match in extensions.conf then the
call is made (or whatever).
This works great because users can dial fast and I don't have to worry about
timeouts or having them press # at the end of their destination number to send
out the digits.
I'm trying to do the same in the SPA8000 units but without any luck. If anyone
is doing something similar with this device then I'd appreciate it if you could
share your relevant config options (dial pattern, etc.).
Thanks!
Vieri
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[asterisk-users] 32 bit server is ok?
hi i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo server no more than 4 to 10 calls at the same time. and a tdm board. waht do you think? thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
You are repeating yourself, paste here sip.conf of each server with register lines AND peer configurations. 2009/1/29 Imanol Pardavila imanol.pardav...@ibercom.com Hi, The SIP messages flow is this: ### AAA.BBB.CCC.DDD: Asterisk 1 IP address EEE.FFF.GGG.HHH: Asterisk 2 IP address ### REGISTER sip:ast2.domain.comSIP/2.0 Via: SIP/2.0/UDP AAA.BBB.CCC.DDD:19646;branch=z9hG4bK0939cc12;rport From: sip:0...@ast2.domain.com sip%3a0...@ast2.domain.com ;tag=as715628d7 To: sip:0...@ast2.domain.com sip%3a0...@ast2.domain.com Call-ID: 13106f3936f01d1c103d5f5230278...@aaa.bbb.ccc.ddd CSeq: 133 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:s...@aaa.bbb.ccc.ddd:19646 Event: registration Content-Length: 0 Using latest REGISTER request as basis request Sending to AAA.BBB.CCC.DDD : 19646 (NAT) Transmitting (NAT) to AAA.BBB.CCC.DDD:19646: SIP/2.0 100 Trying Via: SIP/2.0/UDP AAA.BBB.CCC.DDD:19646;branch=z9hG4bK0939cc12;received=AAA.BBB.CCC.DDD;rport=19646 From: sip:0...@ast2.domain.com sip%3a0...@ast2.domain.com ;tag=as715628d7 To: sip:0...@ast2.domain.com sip%3a0...@ast2.domain.com Call-ID: 13106f3936f01d1c103d5f5230278...@aaa.bbb.ccc.ddd CSeq: 133 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:0...@eee.fff.ggg.hhh Content-Length: 0 Transmitting (NAT) to AAA.BBB.CCC.DDD:19646: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP AAA.BBB.CCC.DDD:19646;branch=z9hG4bK0939cc12;received=AAA.BBB.CCC.DDD;rport=19646 From: sip:0...@ast2.domain.com sip%3a0...@ast2.domain.com ;tag=as715628d7 To: sip:0...@ast2.domain.com sip%3a0...@ast2.domain.com;tag=as5ccb43ac Call-ID: 13106f3936f01d1c103d5f5230278...@aaa.bbb.ccc.ddd CSeq: 133 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7294c1d1 ontent-Length: 0D Asterisk 1 sends an REGISTER without credentials, and Asterisk 2 replies with a 401 message (with Digest algorithm, realm and nonce). I want to configure the Asterisk 1 in order to send REGISTER with credentials. Thanks Regards Grygoriy Dobrovolskyy escribió: Paste your register lines (hide pass) 2009/1/29 Imanol Pardavila imanol.pardav...@ibercom.com mailto:imanol.pardav...@ibercom.com I want to establish a trunk SIP between Asterisk 1 and Asterisk 2, using a sip account (Asterisk 1 acting as a conventional sip user). Thanks Regards Danny Nicholas escribió: Inter-* registry is done with iax.conf, not sip.conf. sip is for phones/sip-lines. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Imanol Pardavila Sent: Thursday, January 29, 2009 10:01 AM To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk Hi, I am trying to register an asterisk (Asterisk 1) against another one (Asterisk 2). My problem is that the REGISTER message goes without credentials and the Asterisk 2 send a 401 message to the Asterisk 1. How can I configure Asterisk 1 to force it to send credentials? I have tried setting Asterisk 2's IP in the realm field of Asterisk's 1 sip.conf, but it doesn`t work. Any ideas? Thanks Regards ___ -- 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:
Re: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
Please check the full syntax for the register= option in sip.conf. You will find it in the section named OUTBOUND SIP REGISTRATIONS of sip.conf.samples in your source code distribution or here for release 1.4: http://svn.digium.com/view/asterisk/branches/1.4/configs/sip.conf.sample?view=co register = user[:secret[:authuse...@host[:port][/extension] Reading documentation is a good thing (TM). /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
Re: [asterisk-users] 32 bit server is ok?
On Thu, Jan 29, 2009 at 04:15:05PM -0200, David fire wrote: hi i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo server no more than 4 to 10 calls at the same time. and a tdm board. Yes. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX
Dear Sir, When trying to send a FAX with T.38I got the following error message [Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on transmission 3058f601-47504...@14.14.14.49 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call 3058f601-47504...@14.14.14.49 - no reply to our critical packet (see doc/sip-retransmit.txt). Regards On Thu, Jan 29, 2009 at 12:04 AM, michel freiha mich...@gmail.com wrote: Dear Danny, Thanks a lot for the help...I'll try and let you know Regards On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas da...@debsinc.comwrote: You need to determine what codecs are expected (sip set debug on from CLI). Commenting out the disallow=all lets * use any available codecs, but may slow down the process or cause undesirable results by using/accounting for unneeded or unwanted codecs. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha *Sent:* Wednesday, January 28, 2009 3:32 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] FAX Dear Sir, What do you mean by manual fax? I need to offer the ability for each extension to use voice and FAX...MAybe the voice will use G729 and the FAX ulaw for the same extension...If I configure the device in a manner that use ulaw for FAX and G729 for voice then this should work smoothly with an extension where G729,ulaw, alaw are allowed? Regards On Wed, Jan 28, 2009 at 11:17 PM, Danny Nicholas da...@debsinc.com wrote: The codecs should only be needed for a manual fax, where a voice interaction might be expected or anticipated. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha *Sent:* Wednesday, January 28, 2009 3:09 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] FAX Dear Sir, If I commant all codecs including disallow=all, then which codec should I define on the extensions from where I'm trying to send FAX? Regards On Wed, Jan 28, 2009 at 10:29 PM, Danny Nicholas da...@debsinc.com wrote: From your sip.conf, you are only allowing ulaw and alaw codes. I'd try adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha *Sent:* Wednesday, January 28, 2009 2:21 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] FAX Dear SIr, please find attached my sip.conf file Regards On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas da...@debsinc.com wrote: Show us your sip.conf -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha *Sent:* Wednesday, January 28, 2009 9:30 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] FAX Hi all, When trying to send a FAX I got the following error: Executing [003228949...@micho:1] Dial(SIP/028949469-08466918, SIP/ 003228949...@80.169.210.181|60) in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949469 -- Couldn't call 0032234534...@1.1.1.1.1 Where I should define the codec other than the extension in order to succeed the call? Regards ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] FAX
Try increasing (or adding) call-limit on sip.conf. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Thursday, January 29, 2009 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear Sir, When trying to send a FAX with T.38I got the following error message [Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on transmission 3058f601-47504...@14.14.14.49 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call 3058f601-47504...@14.14.14.49 - no reply to our critical packet (see doc/sip-retransmit.txt). Regards On Thu, Jan 29, 2009 at 12:04 AM, michel freiha mich...@gmail.com wrote: Dear Danny, Thanks a lot for the help...I'll try and let you know Regards On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas da...@debsinc.com wrote: You need to determine what codecs are expected (sip set debug on from CLI). Commenting out the disallow=all lets * use any available codecs, but may slow down the process or cause undesirable results by using/accounting for unneeded or unwanted codecs. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 3:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear Sir, What do you mean by manual fax? I need to offer the ability for each extension to use voice and FAX...MAybe the voice will use G729 and the FAX ulaw for the same extension...If I configure the device in a manner that use ulaw for FAX and G729 for voice then this should work smoothly with an extension where G729,ulaw, alaw are allowed? Regards On Wed, Jan 28, 2009 at 11:17 PM, Danny Nicholas da...@debsinc.com wrote: The codecs should only be needed for a manual fax, where a voice interaction might be expected or anticipated. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 3:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear Sir, If I commant all codecs including disallow=all, then which codec should I define on the extensions from where I'm trying to send FAX? Regards On Wed, Jan 28, 2009 at 10:29 PM, Danny Nicholas da...@debsinc.com wrote: From your sip.conf, you are only allowing ulaw and alaw codes. I'd try adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear SIr, please find attached my sip.conf file Regards On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas da...@debsinc.com wrote: Show us your sip.conf _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] FAX Hi all, When trying to send a FAX I got the following error: Executing [003228949...@micho:1] Dial(SIP/028949469-08466918, SIP/003228949...@80.169.210.181|60) in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949469 -- Couldn't call 0032234534...@1.1.1.1.1 Where I should define the codec other than the extension in order to succeed the call? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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 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 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 http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] 32 bit server is ok?
The Intel 80386 used 32-bit architecture in 1987...might want to specify make/modelI'm not sure you want to run * on a old Tandy or Packard Bell -jason From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: Thursday, January 29, 2009 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 32 bit server is ok? hi i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo server no more than 4 to 10 calls at the same time. and a tdm board. waht do you think? thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] FAX
Dear Danny, This is the only call on asterisk...:) Regards On Thu, Jan 29, 2009 at 8:35 PM, Danny Nicholas da...@debsinc.com wrote: Try increasing (or adding) call-limit on sip.conf. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha *Sent:* Thursday, January 29, 2009 12:27 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] FAX Dear Sir, When trying to send a FAX with T.38I got the following error message [Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on transmission 3058f601-47504...@14.14.14.49 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call 3058f601-47504...@14.14.14.49 - no reply to our critical packet (see doc/sip-retransmit.txt). Regards On Thu, Jan 29, 2009 at 12:04 AM, michel freiha mich...@gmail.com wrote: Dear Danny, Thanks a lot for the help...I'll try and let you know Regards On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas da...@debsinc.com wrote: You need to determine what codecs are expected (sip set debug on from CLI). Commenting out the disallow=all lets * use any available codecs, but may slow down the process or cause undesirable results by using/accounting for unneeded or unwanted codecs. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha *Sent:* Wednesday, January 28, 2009 3:32 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] FAX Dear Sir, What do you mean by manual fax? I need to offer the ability for each extension to use voice and FAX...MAybe the voice will use G729 and the FAX ulaw for the same extension...If I configure the device in a manner that use ulaw for FAX and G729 for voice then this should work smoothly with an extension where G729,ulaw, alaw are allowed? Regards On Wed, Jan 28, 2009 at 11:17 PM, Danny Nicholas da...@debsinc.com wrote: The codecs should only be needed for a manual fax, where a voice interaction might be expected or anticipated. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha *Sent:* Wednesday, January 28, 2009 3:09 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] FAX Dear Sir, If I commant all codecs including disallow=all, then which codec should I define on the extensions from where I'm trying to send FAX? Regards On Wed, Jan 28, 2009 at 10:29 PM, Danny Nicholas da...@debsinc.com wrote: From your sip.conf, you are only allowing ulaw and alaw codes. I'd try adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha *Sent:* Wednesday, January 28, 2009 2:21 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] FAX Dear SIr, please find attached my sip.conf file Regards On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas da...@debsinc.com wrote: Show us your sip.conf -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha *Sent:* Wednesday, January 28, 2009 9:30 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] FAX Hi all, When trying to send a FAX I got the following error: Executing [003228949...@micho:1] Dial(SIP/028949469-08466918, SIP/ 003228949...@80.169.210.181|60) in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949469 -- Couldn't call 0032234534...@1.1.1.1.1 Where I should define the codec other than the extension in order to succeed the call? Regards ___ -- 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
Re: [asterisk-users] 32 bit server is ok?
i don't know. is like 4 or 5 years old. when they bought it, it was the newest server. thanks 2009/1/29 Jason Aarons (US) jason.aar...@us.didata.com The Intel 80386 used 32-bit architecture in 1987…might want to specify make/model….I'm not sure you want to run * on a old Tandy or Packard Bell -jason *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *David fire *Sent:* Thursday, January 29, 2009 1:15 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] 32 bit server is ok? hi i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo server no more than 4 to 10 calls at the same time. and a tdm board. waht do you think? thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. -- * Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. * ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] FAX
Doesn't matter - the call-limit is important because 1 call can actually be 2-N hops. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Thursday, January 29, 2009 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear Danny, This is the only call on asterisk...:) Regards On Thu, Jan 29, 2009 at 8:35 PM, Danny Nicholas da...@debsinc.com wrote: Try increasing (or adding) call-limit on sip.conf. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Thursday, January 29, 2009 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear Sir, When trying to send a FAX with T.38I got the following error message [Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on transmission 3058f601-47504...@14.14.14.49 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call 3058f601-47504...@14.14.14.49 - no reply to our critical packet (see doc/sip-retransmit.txt). Regards On Thu, Jan 29, 2009 at 12:04 AM, michel freiha mich...@gmail.com wrote: Dear Danny, Thanks a lot for the help...I'll try and let you know Regards On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas da...@debsinc.com wrote: You need to determine what codecs are expected (sip set debug on from CLI). Commenting out the disallow=all lets * use any available codecs, but may slow down the process or cause undesirable results by using/accounting for unneeded or unwanted codecs. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 3:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear Sir, What do you mean by manual fax? I need to offer the ability for each extension to use voice and FAX...MAybe the voice will use G729 and the FAX ulaw for the same extension...If I configure the device in a manner that use ulaw for FAX and G729 for voice then this should work smoothly with an extension where G729,ulaw, alaw are allowed? Regards On Wed, Jan 28, 2009 at 11:17 PM, Danny Nicholas da...@debsinc.com wrote: The codecs should only be needed for a manual fax, where a voice interaction might be expected or anticipated. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 3:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear Sir, If I commant all codecs including disallow=all, then which codec should I define on the extensions from where I'm trying to send FAX? Regards On Wed, Jan 28, 2009 at 10:29 PM, Danny Nicholas da...@debsinc.com wrote: From your sip.conf, you are only allowing ulaw and alaw codes. I'd try adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear SIr, please find attached my sip.conf file Regards On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas da...@debsinc.com wrote: Show us your sip.conf _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] FAX Hi all, When trying to send a FAX I got the following error: Executing [003228949...@micho:1] Dial(SIP/028949469-08466918, SIP/003228949...@80.169.210.181|60) in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949469 -- Couldn't call 0032234534...@1.1.1.1.1 Where I should define the codec other than the extension in order to succeed the call? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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 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] 32 bit server is ok?
If you're not GUI-ing, you could theoretically run * on a 286 since Linux doesn't have the overhead of Windows. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: Thursday, January 29, 2009 12:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 32 bit server is ok? i don't know. is like 4 or 5 years old. when they bought it, it was the newest server. thanks 2009/1/29 Jason Aarons (US) jason.aar...@us.didata.com The Intel 80386 used 32-bit architecture in 1987.might want to specify make/model..I'm not sure you want to run * on a old Tandy or Packard Bell -jason From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: Thursday, January 29, 2009 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 32 bit server is ok? hi i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo server no more than 4 to 10 calls at the same time. and a tdm board. waht do you think? thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. _ Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] FAX
Do you mean call limit on the extension or on the outgoing gateway? Kindly
note that my outbound dialpeer has meeb defined as follow:
[outbound]
exten = _X.,1,Dial(SIP/${ext...@outbound_gw,60)
Regards
On Thu, Jan 29, 2009 at 8:58 PM, Danny Nicholas da...@debsinc.com wrote:
Doesn't matter – the call-limit is important because 1 call can actually
be 2-N hops.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 12:45 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Danny,
This is the only call on asterisk...:)
Regards
On Thu, Jan 29, 2009 at 8:35 PM, Danny Nicholas da...@debsinc.com wrote:
Try increasing (or adding) call-limit on sip.conf.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 12:27 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Sir,
When trying to send a FAX with T.38I got the following error message
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on
transmission 3058f601-47504...@14.14.14.49 for seqno 102 (Critical
Response) -- See doc/sip-retransmit.txt.
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call
3058f601-47504...@14.14.14.49 - no reply to our critical packet (see
doc/sip-retransmit.txt).
Regards
On Thu, Jan 29, 2009 at 12:04 AM, michel freiha mich...@gmail.com wrote:
Dear Danny,
Thanks a lot for the help...I'll try and let you know
Regards
On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas da...@debsinc.com
wrote:
You need to determine what codecs are expected (sip set debug on from
CLI). Commenting out the disallow=all lets * use any available codecs, but
may slow down the process or cause undesirable results by using/accounting
for unneeded or unwanted codecs.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Wednesday, January 28, 2009 3:32 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Sir,
What do you mean by manual fax? I need to offer the ability for each
extension to use voice and FAX...MAybe the voice will use G729 and the FAX
ulaw for the same extension...If I configure the device in a manner that use
ulaw for FAX and G729 for voice then this should work smoothly with an
extension where G729,ulaw, alaw are allowed?
Regards
On Wed, Jan 28, 2009 at 11:17 PM, Danny Nicholas da...@debsinc.com
wrote:
The codecs should only be needed for a manual fax, where a voice
interaction might be expected or anticipated.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Wednesday, January 28, 2009 3:09 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Sir,
If I commant all codecs including disallow=all, then which codec should I
define on the extensions from where I'm trying to send FAX?
Regards
On Wed, Jan 28, 2009 at 10:29 PM, Danny Nicholas da...@debsinc.com
wrote:
From your sip.conf, you are only allowing ulaw and alaw codes. I'd try
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Wednesday, January 28, 2009 2:21 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear SIr,
please find attached my sip.conf file
Regards
On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas da...@debsinc.com wrote:
Show us your sip.conf
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Wednesday, January 28, 2009 9:30 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] FAX
Hi all,
When trying to send a FAX I got the following error:
Executing [003228949...@micho:1] Dial(SIP/028949469-08466918, SIP/
003228949...@80.169.210.181|60) in new stack
[Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format
found to offer. Cancelling call to 003228949469
-- Couldn't call 0032234534...@1.1.1.1.1
Where I should define the codec other than the extension in order to
succeed the call?
Regards
[asterisk-users] Can I use an interact and visa terminal through VoIP?
Hi, Is that reliable? Any known issues? or recommended setups? I am planning on adding the spa2002 devices and attaching the terminal to it. Will this work well? Sincerely, Robert Augustyn ___ -- 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] FAX
On the extension (sip.conf)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Thursday, January 29, 2009 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Do you mean call limit on the extension or on the outgoing gateway? Kindly
note that my outbound dialpeer has meeb defined as follow:
[outbound]
exten = _X.,1,Dial(SIP/${ext...@outbound_gw,60)
Regards
On Thu, Jan 29, 2009 at 8:58 PM, Danny Nicholas da...@debsinc.com wrote:
Doesn't matter - the call-limit is important because 1 call can actually be
2-N hops.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Thursday, January 29, 2009 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear Danny,
This is the only call on asterisk...:)
Regards
On Thu, Jan 29, 2009 at 8:35 PM, Danny Nicholas da...@debsinc.com wrote:
Try increasing (or adding) call-limit on sip.conf.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Thursday, January 29, 2009 12:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear Sir,
When trying to send a FAX with T.38I got the following error message
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on
transmission 3058f601-47504...@14.14.14.49 for seqno 102 (Critical Response)
-- See doc/sip-retransmit.txt.
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call
3058f601-47504...@14.14.14.49 - no reply to our critical packet (see
doc/sip-retransmit.txt).
Regards
On Thu, Jan 29, 2009 at 12:04 AM, michel freiha mich...@gmail.com wrote:
Dear Danny,
Thanks a lot for the help...I'll try and let you know
Regards
On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas da...@debsinc.com wrote:
You need to determine what codecs are expected (sip set debug on from CLI).
Commenting out the disallow=all lets * use any available codecs, but may
slow down the process or cause undesirable results by using/accounting for
unneeded or unwanted codecs.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 3:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear Sir,
What do you mean by manual fax? I need to offer the ability for each
extension to use voice and FAX...MAybe the voice will use G729 and the FAX
ulaw for the same extension...If I configure the device in a manner that use
ulaw for FAX and G729 for voice then this should work smoothly with an
extension where G729,ulaw, alaw are allowed?
Regards
On Wed, Jan 28, 2009 at 11:17 PM, Danny Nicholas da...@debsinc.com wrote:
The codecs should only be needed for a manual fax, where a voice
interaction might be expected or anticipated.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear Sir,
If I commant all codecs including disallow=all, then which codec should I
define on the extensions from where I'm trying to send FAX?
Regards
On Wed, Jan 28, 2009 at 10:29 PM, Danny Nicholas da...@debsinc.com wrote:
From your sip.conf, you are only allowing ulaw and alaw codes. I'd try
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear SIr,
please find attached my sip.conf file
Regards
On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas da...@debsinc.com wrote:
Show us your sip.conf
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] FAX
Hi all,
When trying to send a FAX I got the following error:
Executing [003228949...@micho:1] Dial(SIP/028949469-08466918,
SIP/003228949...@80.169.210.181|60) in new stack
[Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format
found to offer. Cancelling call to 003228949469
-- Couldn't call 0032234534...@1.1.1.1.1
Where I should define the codec other than the extension in order to
Re: [asterisk-users] FAX
I'm getting now the below notice:
rtp.c: Unknown RTP codec 100 received from 'GW address'
On Thu, Jan 29, 2009 at 9:18 PM, michel freiha mich...@gmail.com wrote:
Do you mean call limit on the extension or on the outgoing gateway? Kindly
note that my outbound dialpeer has meeb defined as follow:
[outbound]
exten = _X.,1,Dial(SIP/${ext...@outbound_gw,60)
Regards
On Thu, Jan 29, 2009 at 8:58 PM, Danny Nicholas da...@debsinc.com wrote:
Doesn't matter – the call-limit is important because 1 call can actually
be 2-N hops.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 12:45 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Danny,
This is the only call on asterisk...:)
Regards
On Thu, Jan 29, 2009 at 8:35 PM, Danny Nicholas da...@debsinc.com
wrote:
Try increasing (or adding) call-limit on sip.conf.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 12:27 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Sir,
When trying to send a FAX with T.38I got the following error message
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on
transmission 3058f601-47504...@14.14.14.49 for seqno 102 (Critical
Response) -- See doc/sip-retransmit.txt.
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call
3058f601-47504...@14.14.14.49 - no reply to our critical packet (see
doc/sip-retransmit.txt).
Regards
On Thu, Jan 29, 2009 at 12:04 AM, michel freiha mich...@gmail.com
wrote:
Dear Danny,
Thanks a lot for the help...I'll try and let you know
Regards
On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas da...@debsinc.com
wrote:
You need to determine what codecs are expected (sip set debug on from
CLI). Commenting out the disallow=all lets * use any available codecs, but
may slow down the process or cause undesirable results by using/accounting
for unneeded or unwanted codecs.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Wednesday, January 28, 2009 3:32 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Sir,
What do you mean by manual fax? I need to offer the ability for each
extension to use voice and FAX...MAybe the voice will use G729 and the FAX
ulaw for the same extension...If I configure the device in a manner that use
ulaw for FAX and G729 for voice then this should work smoothly with an
extension where G729,ulaw, alaw are allowed?
Regards
On Wed, Jan 28, 2009 at 11:17 PM, Danny Nicholas da...@debsinc.com
wrote:
The codecs should only be needed for a manual fax, where a voice
interaction might be expected or anticipated.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Wednesday, January 28, 2009 3:09 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Sir,
If I commant all codecs including disallow=all, then which codec should I
define on the extensions from where I'm trying to send FAX?
Regards
On Wed, Jan 28, 2009 at 10:29 PM, Danny Nicholas da...@debsinc.com
wrote:
From your sip.conf, you are only allowing ulaw and alaw codes. I'd try
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Wednesday, January 28, 2009 2:21 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear SIr,
please find attached my sip.conf file
Regards
On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas da...@debsinc.com
wrote:
Show us your sip.conf
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Wednesday, January 28, 2009 9:30 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] FAX
Hi all,
When trying to send a FAX I got the following error:
Executing [003228949...@micho:1] Dial(SIP/028949469-08466918, SIP/
003228949...@80.169.210.181|60) in new stack
[Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio
format found to offer. Cancelling
Re: [asterisk-users] Asterisk - Trixbox
Should Trixbox be sending calls to the s extension in the first place? I can't set an s extension because there are many independent phone numbers in that context that worked fine before my provider switched to Trixbox. Also, why would the 8159093011 phone number be showing up in the sip debugging when that number isn't even present on that machine? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Adrià Vidal adriavi...@gmail.com Sent: Friday, January 16, 2009 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett asterisk-us...@ics-il.net wrote: My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com I think you need something inside [DID-incoming] like for example... exten = s,1,NoOP(-incoming call---) exten = s,n,Playback(wellcome) # Looking for s in DID-incoming (domain 208.100.1.33) # Reliably Transmitting (no NAT) to 208.1.87.235:5060: # SIP/2.0 404 Not Found -- -- Adrià Vidal adriavi...@gmail.com ___ -- 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] FAX
Good new is your'e getting somewhere. Bad new is - you have to modify
rtp.c to allow this codec. You should be able to duplicate a line (around
1390) and change the value from
[34] = {1, AST_FORMAT_H263},
To
[100] = {1, AST_FORMAT_H100},
Then just do a make make install on asterisk again.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Thursday, January 29, 2009 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
I'm getting now the below notice:
rtp.c: Unknown RTP codec 100 received from 'GW address'
On Thu, Jan 29, 2009 at 9:18 PM, michel freiha mich...@gmail.com wrote:
Do you mean call limit on the extension or on the outgoing gateway? Kindly
note that my outbound dialpeer has meeb defined as follow:
[outbound]
exten = _X.,1,Dial(SIP/${ext...@outbound_gw,60)
Regards
On Thu, Jan 29, 2009 at 8:58 PM, Danny Nicholas da...@debsinc.com wrote:
Doesn't matter - the call-limit is important because 1 call can actually be
2-N hops.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Thursday, January 29, 2009 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear Danny,
This is the only call on asterisk...:)
Regards
On Thu, Jan 29, 2009 at 8:35 PM, Danny Nicholas da...@debsinc.com wrote:
Try increasing (or adding) call-limit on sip.conf.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Thursday, January 29, 2009 12:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear Sir,
When trying to send a FAX with T.38I got the following error message
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on
transmission 3058f601-47504...@14.14.14.49 for seqno 102 (Critical Response)
-- See doc/sip-retransmit.txt.
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call
3058f601-47504...@14.14.14.49 - no reply to our critical packet (see
doc/sip-retransmit.txt).
Regards
On Thu, Jan 29, 2009 at 12:04 AM, michel freiha mich...@gmail.com wrote:
Dear Danny,
Thanks a lot for the help...I'll try and let you know
Regards
On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas da...@debsinc.com wrote:
You need to determine what codecs are expected (sip set debug on from CLI).
Commenting out the disallow=all lets * use any available codecs, but may
slow down the process or cause undesirable results by using/accounting for
unneeded or unwanted codecs.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 3:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear Sir,
What do you mean by manual fax? I need to offer the ability for each
extension to use voice and FAX...MAybe the voice will use G729 and the FAX
ulaw for the same extension...If I configure the device in a manner that use
ulaw for FAX and G729 for voice then this should work smoothly with an
extension where G729,ulaw, alaw are allowed?
Regards
On Wed, Jan 28, 2009 at 11:17 PM, Danny Nicholas da...@debsinc.com wrote:
The codecs should only be needed for a manual fax, where a voice
interaction might be expected or anticipated.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear Sir,
If I commant all codecs including disallow=all, then which codec should I
define on the extensions from where I'm trying to send FAX?
Regards
On Wed, Jan 28, 2009 at 10:29 PM, Danny Nicholas da...@debsinc.com wrote:
From your sip.conf, you are only allowing ulaw and alaw codes. I'd try
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear SIr,
please find attached my sip.conf file
Regards
On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas da...@debsinc.com wrote:
Show us your sip.conf
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Re: [asterisk-users] Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
On Thu, 29 Jan 2009, Thomas Stein wrote: On Thursday 29 January 2009 09:23:41 Remco Barendse wrote: 1.4.23.1 doesn't seem to work for me. I did an in-place upgrade of Asterisk 1.4.21 and upgraded to the latest zaptel as well. Incoming calls stopped working. Whenever an extension was trying to pickup the phone by doing a group pickup with *8 the extension just got dead audio and the next phone in the group stared ringing. Yeah. Thats http://bugs.digium.com:80/view.php?id=14206 I'm also concerned about that one: http://bugs.digium.com:80/view.php?id=13488 cheers t. Thanks for your reply, indeed that is the problem. Strange that this stable release is still prominently on the asterisk.org website as the latest and greatest. The latest bug you mentioned is only valid for mISDN installations i think? Thanks again! ___ -- 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] FAX
Dear Danny,
i got the following error during make
[CC] rtp.c - rtp.o
rtp.c:1390:3: error: invalid preprocessing directive #[
rtp.c:1391: error: ‘AST_FORMAT_H100’ undeclared here (not in a function)
rtp.c:1392: error: expected ‘}’ before ‘[’ token
make[1]: *** [rtp.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-1.4.22.1/main'
make: *** [main] Error 2
regards
On Thu, Jan 29, 2009 at 9:48 PM, Danny Nicholas da...@debsinc.com wrote:
Good new is your'e getting somewhere. Bad new is – you have to modify
rtp.c to allow this codec. You should be able to duplicate a line (around
1390) and change the value from
[34] = {1, AST_FORMAT_H263},
To
[100] = {1, AST_FORMAT_H100},
Then just do a make make install on asterisk again.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 1:35 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
I'm getting now the below notice:
rtp.c: Unknown RTP codec 100 received from 'GW address'
On Thu, Jan 29, 2009 at 9:18 PM, michel freiha mich...@gmail.com wrote:
Do you mean call limit on the extension or on the outgoing gateway? Kindly
note that my outbound dialpeer has meeb defined as follow:
[outbound]
exten = _X.,1,Dial(SIP/${ext...@outbound_gw,60)
Regards
On Thu, Jan 29, 2009 at 8:58 PM, Danny Nicholas da...@debsinc.com wrote:
Doesn't matter – the call-limit is important because 1 call can actually be
2-N hops.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 12:45 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Danny,
This is the only call on asterisk...:)
Regards
On Thu, Jan 29, 2009 at 8:35 PM, Danny Nicholas da...@debsinc.com wrote:
Try increasing (or adding) call-limit on sip.conf.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 12:27 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Sir,
When trying to send a FAX with T.38I got the following error message
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on
transmission 3058f601-47504...@14.14.14.49 for seqno 102 (Critical
Response) -- See doc/sip-retransmit.txt.
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call
3058f601-47504...@14.14.14.49 - no reply to our critical packet (see
doc/sip-retransmit.txt).
Regards
On Thu, Jan 29, 2009 at 12:04 AM, michel freiha mich...@gmail.com wrote:
Dear Danny,
Thanks a lot for the help...I'll try and let you know
Regards
On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas da...@debsinc.com
wrote:
You need to determine what codecs are expected (sip set debug on from
CLI). Commenting out the disallow=all lets * use any available codecs, but
may slow down the process or cause undesirable results by using/accounting
for unneeded or unwanted codecs.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Wednesday, January 28, 2009 3:32 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Sir,
What do you mean by manual fax? I need to offer the ability for each
extension to use voice and FAX...MAybe the voice will use G729 and the FAX
ulaw for the same extension...If I configure the device in a manner that use
ulaw for FAX and G729 for voice then this should work smoothly with an
extension where G729,ulaw, alaw are allowed?
Regards
On Wed, Jan 28, 2009 at 11:17 PM, Danny Nicholas da...@debsinc.com
wrote:
The codecs should only be needed for a manual fax, where a voice
interaction might be expected or anticipated.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Wednesday, January 28, 2009 3:09 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Sir,
If I commant all codecs including disallow=all, then which codec should I
define on the extensions from where I'm trying to send FAX?
Regards
On Wed, Jan 28, 2009 at 10:29 PM, Danny Nicholas da...@debsinc.com
wrote:
From your sip.conf, you are only allowing ulaw and alaw codes. I'd try
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw
Re: [asterisk-users] FAX
[3] = {1, AST_FORMAT_GSM},
[4] = {1, AST_FORMAT_G723_1},
[5] = {1, AST_FORMAT_ADPCM}, /* 8 kHz */
[6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */
[7] = {1, AST_FORMAT_LPC10},
[8] = {1, AST_FORMAT_ALAW},
[9] = {1, AST_FORMAT_G722},
[10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
[11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
[13] = {0, AST_RTP_CN},
[16] = {1, AST_FORMAT_ADPCM}, /* 11.025 kHz */
[17] = {1, AST_FORMAT_ADPCM}, /* 22.050 kHz */
[18] = {1, AST_FORMAT_G729A},
[19] = {0, AST_RTP_CN}, /* Also used for CN */
[26] = {1, AST_FORMAT_JPEG},
[31] = {1, AST_FORMAT_H261},
# [34] = {1, AST_FORMAT_H263},
[100] = {1, AST_FORMAT_H100}
[103] = {1, AST_FORMAT_H263_PLUS},
[97] = {1, AST_FORMAT_ILBC},
[99] = {1, AST_FORMAT_H264},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},
[111] = {1, AST_FORMAT_G726},
[112] = {1, AST_FORMAT_G726_AAL2},
[121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
On Thu, Jan 29, 2009 at 9:48 PM, Danny Nicholas da...@debsinc.com wrote:
Good new is your'e getting somewhere. Bad new is – you have to modify
rtp.c to allow this codec. You should be able to duplicate a line (around
1390) and change the value from
[34] = {1, AST_FORMAT_H263},
To
[100] = {1, AST_FORMAT_H100},
Then just do a make make install on asterisk again.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 1:35 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
I'm getting now the below notice:
rtp.c: Unknown RTP codec 100 received from 'GW address'
On Thu, Jan 29, 2009 at 9:18 PM, michel freiha mich...@gmail.com wrote:
Do you mean call limit on the extension or on the outgoing gateway? Kindly
note that my outbound dialpeer has meeb defined as follow:
[outbound]
exten = _X.,1,Dial(SIP/${ext...@outbound_gw,60)
Regards
On Thu, Jan 29, 2009 at 8:58 PM, Danny Nicholas da...@debsinc.com wrote:
Doesn't matter – the call-limit is important because 1 call can actually be
2-N hops.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 12:45 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Danny,
This is the only call on asterisk...:)
Regards
On Thu, Jan 29, 2009 at 8:35 PM, Danny Nicholas da...@debsinc.com wrote:
Try increasing (or adding) call-limit on sip.conf.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 12:27 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Sir,
When trying to send a FAX with T.38I got the following error message
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on
transmission 3058f601-47504...@14.14.14.49 for seqno 102 (Critical
Response) -- See doc/sip-retransmit.txt.
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call
3058f601-47504...@14.14.14.49 - no reply to our critical packet (see
doc/sip-retransmit.txt).
Regards
On Thu, Jan 29, 2009 at 12:04 AM, michel freiha mich...@gmail.com wrote:
Dear Danny,
Thanks a lot for the help...I'll try and let you know
Regards
On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas da...@debsinc.com
wrote:
You need to determine what codecs are expected (sip set debug on from
CLI). Commenting out the disallow=all lets * use any available codecs, but
may slow down the process or cause undesirable results by using/accounting
for unneeded or unwanted codecs.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Wednesday, January 28, 2009 3:32 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Sir,
What do you mean by manual fax? I need to offer the ability for each
extension to use voice and FAX...MAybe the voice will use G729 and the FAX
ulaw for the same extension...If I configure the device in a manner that use
ulaw for FAX and G729 for voice then this should work smoothly with an
extension where G729,ulaw, alaw are allowed?
Regards
On Wed, Jan 28, 2009 at 11:17 PM, Danny Nicholas da...@debsinc.com
wrote:
The codecs should only be needed for a manual fax, where a voice
interaction might be expected or anticipated.
Re: [asterisk-users] 32 bit server is ok?
On 16:15, Thu 29 Jan 09, David fire wrote: hi i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo server no more than 4 to 10 calls at the same time. and a tdm board. waht do you think? Yup, no problem. Any X86 machine with a pci slot will be able to handle this. even 486 machines will be able to do that. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados 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] 32 bit server is ok?
On Thu, 29 Jan 2009, Danny Nicholas wrote: If you're not GUI-ing, you could theoretically run * on a 286 since Linux doesn't have the overhead of Windows. According to Wikipedia's entry for Intel 80286 After the 6 and 8 MHz initial releases, it was subsequently scaled up to 12.5 MHz. (AMD and Harris later pushed the architecture to speeds as high as 20 MHz and 25 MHz, respectively.) On average, the 80286 had a speed of about 0.21 instructions per clock. [2] The 6 MHz model operated at 0.9 MIPS, the 10 MHz model at 1.5 MIPS, and the 12 MHz model at 1.8 MIPs. and later Having a 24-bit address bus, the 286 was able to address up to 16 MB of RAM... Theoretically? I run Asterisk on a 286. Yep. It receives faxes using GSM all day long. Yep. That's what I do :) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX
Got too cute. Make AST_FORMAT_H100 be AST_FORMAT_H263.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Thursday, January 29, 2009 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear Danny,
i got the following error during make
[CC] rtp.c - rtp.o
rtp.c:1390:3: error: invalid preprocessing directive #[
rtp.c:1391: error: ‘AST_FORMAT_H100’ undeclared here (not in a function)
rtp.c:1392: error: expected ‘}’ before ‘[’ token
make[1]: *** [rtp.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-1.4.22.1/main'
make: *** [main] Error 2
regards
On Thu, Jan 29, 2009 at 9:48 PM, Danny Nicholas da...@debsinc.com wrote:
Good new is your'e getting somewhere. Bad new is – you have to modify rtp.c
to allow this codec. You should be able to duplicate a line (around 1390) and
change the value from
[34] = {1, AST_FORMAT_H263},
To
[100] = {1, AST_FORMAT_H100},
Then just do a make make install on asterisk again.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Thursday, January 29, 2009 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
I'm getting now the below notice:
rtp.c: Unknown RTP codec 100 received from 'GW address'
On Thu, Jan 29, 2009 at 9:18 PM, michel freiha mich...@gmail.com wrote:
Do you mean call limit on the extension or on the outgoing gateway? Kindly note
that my outbound dialpeer has meeb defined as follow:
[outbound]
exten = _X.,1,Dial(SIP/${ext...@outbound_gw,60)
Regards
On Thu, Jan 29, 2009 at 8:58 PM, Danny Nicholas da...@debsinc.com wrote:
Doesn't matter – the call-limit is important because 1 call can actually be 2-N
hops.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Thursday, January 29, 2009 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear Danny,
This is the only call on asterisk...:)
Regards
On Thu, Jan 29, 2009 at 8:35 PM, Danny Nicholas da...@debsinc.com wrote:
Try increasing (or adding) call-limit on sip.conf.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Thursday, January 29, 2009 12:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear Sir,
When trying to send a FAX with T.38I got the following error message
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on
transmission 3058f601-47504...@14.14.14.49 for seqno 102 (Critical Response) --
See doc/sip-retransmit.txt.
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call
3058f601-47504...@14.14.14.49 - no reply to our critical packet (see
doc/sip-retransmit.txt).
Regards
On Thu, Jan 29, 2009 at 12:04 AM, michel freiha mich...@gmail.com wrote:
Dear Danny,
Thanks a lot for the help...I'll try and let you know
Regards
On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas da...@debsinc.com wrote:
You need to determine what codecs are expected (sip set debug on from CLI).
Commenting out the disallow=all lets * use any available codecs, but may slow
down the process or cause undesirable results by using/accounting for unneeded
or unwanted codecs.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 3:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear Sir,
What do you mean by manual fax? I need to offer the ability for each extension
to use voice and FAX...MAybe the voice will use G729 and the FAX ulaw for the
same extension...If I configure the device in a manner that use ulaw for FAX
and G729 for voice then this should work smoothly with an extension where
G729,ulaw, alaw are allowed?
Regards
On Wed, Jan 28, 2009 at 11:17 PM, Danny Nicholas da...@debsinc.com wrote:
The codecs should only be needed for a manual fax, where a voice interaction
might be expected or anticipated.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear Sir,
If I commant all codecs including disallow=all, then which codec should I
define on the extensions from where I'm trying to send FAX?
Regards
On Wed, Jan 28, 2009 at 10:29 PM, Danny Nicholas da...@debsinc.com wrote:
From your sip.conf, you are only allowing ulaw and
Re: [asterisk-users] 32 bit server is ok?
2009/1/29 Tzafrir Cohen tzafrir.co...@xorcom.com On Thu, Jan 29, 2009 at 04:15:05PM -0200, David fire wrote: hi i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo server no more than 4 to 10 calls at the same time. and a tdm board. Yes. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for the fast answer -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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 Recording Alias
David @ULC schrieb: Where did I make mistake ? You posted (even re-posted) a question about Vicidial and Apache configuration on asterisk-users. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
On Thursday 29 January 2009 13:50:19 Remco Barendse wrote: On Thu, 29 Jan 2009, Thomas Stein wrote: On Thursday 29 January 2009 09:23:41 Remco Barendse wrote: 1.4.23.1 doesn't seem to work for me. I did an in-place upgrade of Asterisk 1.4.21 and upgraded to the latest zaptel as well. Incoming calls stopped working. Whenever an extension was trying to pickup the phone by doing a group pickup with *8 the extension just got dead audio and the next phone in the group stared ringing. Yeah. Thats http://bugs.digium.com:80/view.php?id=14206 I'm also concerned about that one: http://bugs.digium.com:80/view.php?id=13488 cheers t. Thanks for your reply, indeed that is the problem. Strange that this stable release is still prominently on the asterisk.org website as the latest and greatest. Where do you see us denoting any release as stable (or defining what that term actually means)? We release when we think that we've eliminated the bugs we can find, and then people find more bugs. If you can fix bugs before they're reported, we'd love to have you contribute to the development effort. -- Tilghman ___ -- 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 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
For a while we were seeing RC(release cantidates) release announcements and I can see that there were RC release for this 1.4.23 release. Any reason they aren't being publicized, or am I just looking in the wrong place? We always do compatibility testing before putting a new release in production and at this point 1.4.21.2 is the most recent stable release as far as we are concerned. Of course 1.4 wasn't really stable until 1.4.18, which is when the RC releases started too. MATT--- On 1/29/09, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Thursday 29 January 2009 13:50:19 Remco Barendse wrote: On Thu, 29 Jan 2009, Thomas Stein wrote: On Thursday 29 January 2009 09:23:41 Remco Barendse wrote: 1.4.23.1 doesn't seem to work for me. I did an in-place upgrade of Asterisk 1.4.21 and upgraded to the latest zaptel as well. Incoming calls stopped working. Whenever an extension was trying to pickup the phone by doing a group pickup with *8 the extension just got dead audio and the next phone in the group stared ringing. Yeah. Thats http://bugs.digium.com:80/view.php?id=14206 I'm also concerned about that one: http://bugs.digium.com:80/view.php?id=13488 cheers t. Thanks for your reply, indeed that is the problem. Strange that this stable release is still prominently on the asterisk.org website as the latest and greatest. Where do you see us denoting any release as stable (or defining what that term actually means)? We release when we think that we've eliminated the bugs we can find, and then people find more bugs. If you can fix bugs before they're reported, we'd love to have you contribute to the development effort. -- Tilghman ___ -- 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] FAX
I did but get the following erro on SIP:
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 12.12.12.12:16444
Found audio description format PCMU for ID 0
Found unknown media description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing),
combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 12.12.12.12:16444
set_destination: Parsing sip:johan...@12.12.12.12:5060 for address/port to
send to
set_destination: set destination to 12.12.12.12, port 5060
Transmitting (NAT) to 84.198.68.243:43996:
On Thu, Jan 29, 2009 at 10:45 PM, Danny Nicholas da...@debsinc.com wrote:
Got too cute. Make AST_FORMAT_H100 be AST_FORMAT_H263.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 2:10 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Danny,
i got the following error during make
[CC] rtp.c - rtp.o
rtp.c:1390:3: error: invalid preprocessing directive #[
rtp.c:1391: error: ‘AST_FORMAT_H100’ undeclared here (not in a
function)
rtp.c:1392: error: expected ‘}’ before ‘[’ token
make[1]: *** [rtp.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-1.4.22.1/main'
make: *** [main] Error 2
regards
On Thu, Jan 29, 2009 at 9:48 PM, Danny Nicholas da...@debsinc.com wrote:
Good new is your'e getting somewhere. Bad new is – you have to modify
rtp.c to allow this codec. You should be able to duplicate a line (around
1390) and change the value from
[34] = {1, AST_FORMAT_H263},
To
[100] = {1, AST_FORMAT_H100},
Then just do a make make install on asterisk again.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 1:35 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
I'm getting now the below notice:
rtp.c: Unknown RTP codec 100 received from 'GW address'
On Thu, Jan 29, 2009 at 9:18 PM, michel freiha mich...@gmail.com wrote:
Do you mean call limit on the extension or on the outgoing gateway? Kindly
note that my outbound dialpeer has meeb defined as follow:
[outbound]
exten = _X.,1,Dial(SIP/${ext...@outbound_gw,60)
Regards
On Thu, Jan 29, 2009 at 8:58 PM, Danny Nicholas da...@debsinc.com wrote:
Doesn't matter – the call-limit is important because 1 call can actually be
2-N hops.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 12:45 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Danny,
This is the only call on asterisk...:)
Regards
On Thu, Jan 29, 2009 at 8:35 PM, Danny Nicholas da...@debsinc.com wrote:
Try increasing (or adding) call-limit on sip.conf.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Thursday, January 29, 2009 12:27 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Sir,
When trying to send a FAX with T.38I got the following error message
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on
transmission 3058f601-47504...@14.14.14.49 for seqno 102 (Critical
Response) -- See doc/sip-retransmit.txt.
[Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call
3058f601-47504...@14.14.14.49 - no reply to our critical packet (see
doc/sip-retransmit.txt).
Regards
On Thu, Jan 29, 2009 at 12:04 AM, michel freiha mich...@gmail.com wrote:
Dear Danny,
Thanks a lot for the help...I'll try and let you know
Regards
On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas da...@debsinc.com
wrote:
You need to determine what codecs are expected (sip set debug on from
CLI). Commenting out the disallow=all lets * use any available codecs, but
may slow down the process or cause undesirable results by using/accounting
for unneeded or unwanted codecs.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Wednesday, January 28, 2009 3:32 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] FAX
Dear Sir,
What do you mean by manual fax? I need to offer the
Re: [asterisk-users] Asterisk 1.2.31.1, 1.4 .22.2, 1.4.23.1, and 1.6.0.5 released
On Thursday 29 January 2009 16:00:14 Matt Florell wrote: For a while we were seeing RC(release cantidates) release announcements and I can see that there were RC release for this 1.4.23 release. Any reason they aren't being publicized, or am I just looking in the wrong place? http://lists.digium.com/pipermail/asterisk-users/2008-December/222727.html http://lists.digium.com/pipermail/asterisk-users/2008-December/223668.html http://lists.digium.com/pipermail/asterisk-users/2009-January/224940.html I'd say you just missed them, as they were published to this list, as evidenced by the archives. -- Tilghman ___ -- 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] Can I use an interact and visa terminal through VoIP?
Robert Augustyn schrieb: Hi, Is that reliable? Any known issues? or recommended setups? I am planning on adding the spa2002 devices and attaching the terminal to it. Will this work well? Sincerely, Robert Augustyn hello, my expierince with data connections like Modem over voip is just to take the hands of it. when doing this you should disable all echo cancelation on the ata and use a high quality codec like g711u or g711a, but i think you wont be happy with this. best regards. steve smith ___ -- 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] Can I use an interact and visa terminal through VoIP?
Many modern credit card authorization terminals can use an Internet, rather than phone, connection. That tends to be quicker, as well as being more reliable. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan Schmidt Sent: Thursday, January 29, 2009 4:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can I use an interact and visa terminal through VoIP? Robert Augustyn schrieb: Hi, Is that reliable? Any known issues? or recommended setups? I am planning on adding the spa2002 devices and attaching the terminal to it. Will this work well? Sincerely, Robert Augustyn hello, my expierince with data connections like Modem over voip is just to take the hands of it. when doing this you should disable all echo cancelation on the ata and use a high quality codec like g711u or g711a, but i think you wont be happy with this. best regards. steve smith ___ -- 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] Callback / Camp / Extention Free notify?
Quick solution that comes into mind:
Set(exten_copy = ${EXTEN});
Dial(SIP/${EXTEN})
if (${DIALSTATUS}=BUSY) {
// prompt for camp
Set(DB(camp/${EXTEN}/call_to)=${CALLERID(num));
}
h = {
Set(call_to=${DB(camp/${exten_copy}/call_to)});
if (${call_to}!=) {
Set(DB(camp/${exten_copy}/call_to)=);
System(call_to ${exten_copy} ${call_to});
}
}
So, in case if phone2 is busy, store callerid of phone1 in database,
so when phone2 will hangup it will triger a script call_to which
however can originate call trough manager or call-file.
Of course you will need some additional handling in case if multiple
callers decide to camp, or diferent protocols are used, etc.
You could call a batch script from the dialplan that parses the output
of 'show hints' with a simple grep to find the status of the individual
in question.
PaulH
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Re: [asterisk-users] Callback / Camp / Extention Free notify?
Yes, the more expensive ones do. The majority do not. Linksys phones. I have a Snom 320 and an Aastra 480i on my desk, and one of the reasons I love them (especially the Aastra) is the BLF features. Its not so much knowing if the user is busy or not, its the ability to be automatically notified once the user becomes available. * *No problem - it will be doable, it's just how much effort will be needed to get it working, and then how much more effort to get it 'perfect'. I know that a lot of people have been through exactly what you are going through with regards to legacy features - I had to write a piece of dialplan code to return blind transfers back to the person who started the transfer if the extension they were calling did not answer...just like the old phone system they had...because attended transfers were too hard. later, PaulH ** ___ -- 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] manager API with no login?
I've been searching around for a while, and haven't found an answer to this question, so here goes: Does anyone know if AMI can be configured to allow requests from another client without having to authenticate first? I would like to be able to restrict it based on IP address, and not require a login. Any help is appreciated. Thanks! -- Brooks R. Bridges Telecommunications Manager Ifbyphone, Inc. Phone: (847) 983-3000 Fax: (847) 676-6553 bbrid...@ifbyphone.com http://www.ifbyphone.com ___ -- 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 1.6.1 Release Candidate 1 Now Available
The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 1.6.1, tagged as version 1.6.1-rc1. Release candidate 1.6.1-rc1 is available for immediate download at http://downloads.digium.com/. This release candidate includes a fix to SIP registrations when using realtime. Additional crash issues have been resolved, in addition to making chan_sip more robust in handling unique scenarios. Issues found in this release candidate can be reported at http://bugs.digium.com/. For a full list of changes in this release candidate, please see the ChangeLog: http://svn.digium.com/view/asterisk/tags/1.6.1-rc1/ChangeLog?view=co Also see the CHANGES file for useful information about what is new in Asterisk 1.6.1. See the CHANGES file at: http://svn.digium.com/view/asterisk/tags/1.6.1-rc1/CHANGES?view=co Thank you for your continued support of Asterisk! ___ -- 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] manager API with no login?
On 30/01/2009 12:55 p.m., Brooks Bridges wrote: I've been searching around for a while, and haven't found an answer to this question, so here goes: Does anyone know if AMI can be configured to allow requests from another client without having to authenticate first? I would like to be able to restrict it based on IP address, and not require a login. No it cannot and should not. If you require this you may want to proxy the communication. Bear in mind that you cannot always trust the source ip (you can work around this by blocking external connections from ip addresses used by the internal machines). -- 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) ___ -- 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 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
Yep, my bad I found them once I searched with the dash '-' after the 1.4.23. They were lost in the flood of users list mail in my inbox. I wonder if these could also be posted on the asterisk-announce list more consistently? I see a few releases on the announce list, but last 1.4 one was December 2nd and nothing after that on that list except for a few vulnerability postings. I know it would help me to get those release notices on that list, then I could flag them better so my mail viewer will smack me on the head to read them when they come in. Thanks, MATT--- On 1/29/09, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Thursday 29 January 2009 16:00:14 Matt Florell wrote: For a while we were seeing RC(release cantidates) release announcements and I can see that there were RC release for this 1.4.23 release. Any reason they aren't being publicized, or am I just looking in the wrong place? http://lists.digium.com/pipermail/asterisk-users/2008-December/222727.html http://lists.digium.com/pipermail/asterisk-users/2008-December/223668.html http://lists.digium.com/pipermail/asterisk-users/2009-January/224940.html I'd say you just missed them, as they were published to this list, as evidenced by the archives. -- Tilghman ___ -- 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] Scope of variable
On Wed, Jan 28, 2009 at 2:48 PM, Jim Dickenson dicken...@cfmc.com wrote: I have this extension: exten = 1322,n,Playback(tt-weasels) Clearly the problem is that weasels have eaten [your] phone system. :-p But really, you posted your 1322 condition, and then asked about something happening with 1321. So you need to take a better look at your extensions.conf and find out what is happening somewhere else. Adding on variables you're not defining in your sample doesn't make it any easier. All I can say is that the 1322 condition is likely not the problem, or at least not the problem you're asking about here. Try commenting it out entirely, reload your dialplan, and do process of elimination until you find the section of your dialplan you're actually having a problem with. You might also like to enable heavier debugging and confirm that when you set those variables they end up with the values you think they get, or your if's are going to eval differently, blah, blah, blah. ___ -- 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] looking for a link or pdf ot something about opensip/openser and load balancing
hi i need a link or something about asterisk load balancing i cant find any, i only found a paragraf in an email anything wiil be wolcome thanks! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
Has anybody ever asked Digium to provide something like the kernel.org RSS feed? My impression was this was created because kernel.org was tired of how many people built cron-ified wget scripts against kernel.org | diff against last get | sendmail you get the idea Perhaps downloads.digium.com has the same problem? the kernel.org rss feed is: http://www.kernel.org/kdist/rss.xml On Thu, Jan 29, 2009 at 8:44 PM, Matt Florell astma...@gmail.com wrote: Yep, my bad I found them once I searched with the dash '-' after the 1.4.23. They were lost in the flood of users list mail in my inbox. I wonder if these could also be posted on the asterisk-announce list more consistently? I see a few releases on the announce list, but last 1.4 one was December 2nd and nothing after that on that list except for a few vulnerability postings. I know it would help me to get those release notices on that list, then I could flag them better so my mail viewer will smack me on the head to read them when they come in. Thanks, MATT--- On 1/29/09, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Thursday 29 January 2009 16:00:14 Matt Florell wrote: For a while we were seeing RC(release cantidates) release announcements and I can see that there were RC release for this 1.4.23 release. Any reason they aren't being publicized, or am I just looking in the wrong place? http://lists.digium.com/pipermail/asterisk-users/2008-December/222727.html http://lists.digium.com/pipermail/asterisk-users/2008-December/223668.html http://lists.digium.com/pipermail/asterisk-users/2009-January/224940.html I'd say you just missed them, as they were published to this list, as evidenced by the archives. -- Tilghman ___ -- 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] looking for a link or pdf ot something about opensip/openser and load balancing
http://www.kamailio.org/docs/modules/1.4.x/dispatcher.html David fire wrote: hi i need a link or something about asterisk load balancing i cant find any, i only found a paragraf in an email anything wiil be wolcome thanks! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] looking for a link or pdf ot something about opensip/openser and load balancing
http://lists.digium.com/mailman/listinfo/ consider joining asterisk-ha-clustering or at least looking through their archives I do all my balancing via SIP and DNS round robin (and a few other more custom things). Works quite well for me. On Thu, Jan 29, 2009 at 8:56 PM, David fire ddf...@gmail.com wrote: hi i need a link or something about asterisk load balancing i cant find any, i only found a paragraf in an email anything wiil be wolcome thanks! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
David Backeberg schrieb: Has anybody ever asked Digium to provide something like the kernel.org RSS feed? My impression was this was created because kernel.org was tired of how many people built cron-ified wget scripts against kernel.org | diff against last get | sendmail you get the idea Perhaps downloads.digium.com has the same problem? the kernel.org rss feed is: http://www.kernel.org/kdist/rss.xml http://www.kempgen.net/asterisk/current/ has an RSS feed (and a release timeline). e.g. http://www.kempgen.net/asterisk/current/version/asterisk-1.4 always points to the current 1.4.x release, http://www.kempgen.net/asterisk/current/version/asterisk-1.6 always points to the current 1.6.x release etc. Release candidates (-rc) are not considered. Only now I realize that new hotfix releases (1.4.22.2) are not considered if a release with a higher version (1.4.23.1) is already in the database. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] 32 bit server is ok?
Nope - 286 didn't have protected memory access mode, which is key to *nix kernels. j On Thu, 29 Jan 2009, Danny Nicholas wrote: If you're not GUI-ing, you could theoretically run * on a 286 since Linux doesn't have the overhead of Windows. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: Thursday, January 29, 2009 12:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 32 bit server is ok? i don't know. is like 4 or 5 years old. when they bought it, it was the newest server. thanks 2009/1/29 Jason Aarons (US) jason.aar...@us.didata.com The Intel 80386 used 32-bit architecture in 1987.might want to specify make/model..I'm not sure you want to run * on a old Tandy or Packard Bell -jason From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: Thursday, January 29, 2009 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 32 bit server is ok? hi i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo server no more than 4 to 10 calls at the same time. and a tdm board. waht do you think? thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. _ Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] 32 bit server is ok?
This thread made me nostalgic - see this: http://en.wikipedia.org/wiki/MINIX I took a course based on MINIX (as did Linus Torvald) back in 1989 and recall building symbolic links into its kernel as part of a class project. On a 386SX I built in my dorm room. j On Fri, 30 Jan 2009, Jeff LaCoursiere wrote: Nope - 286 didn't have protected memory access mode, which is key to *nix kernels. j On Thu, 29 Jan 2009, Danny Nicholas wrote: If you're not GUI-ing, you could theoretically run * on a 286 since Linux doesn't have the overhead of Windows. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: Thursday, January 29, 2009 12:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 32 bit server is ok? i don't know. is like 4 or 5 years old. when they bought it, it was the newest server. thanks 2009/1/29 Jason Aarons (US) jason.aar...@us.didata.com The Intel 80386 used 32-bit architecture in 1987.might want to specify make/model..I'm not sure you want to run * on a old Tandy or Packard Bell -jason From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: Thursday, January 29, 2009 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 32 bit server is ok? hi i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo server no more than 4 to 10 calls at the same time. and a tdm board. waht do you think? thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. _ Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] manager API with no login?
On Thu, Jan 29, 2009 at 05:55:39PM -0600, Brooks Bridges wrote: I've been searching around for a while, and haven't found an answer to this question, so here goes: Does anyone know if AMI can be configured to allow requests from another client without having to authenticate first? I would like to be able to restrict it based on IP address, and not require a login. You can easily do that yourself with a proxy server. Is this a good idea? not sure. One place where quite a similar approach is used: http://monast.sourceforge.net/ -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Where to find db1_dump185 in debian packages ?
Hello, Here http://www.voip-info.org/wiki/view/Asterisk+database , you can read: Also, since it's a normal Berkely db1 (version185) file its contents can be viewed/dumped with the standard *db1_dump185* tool. Thus db1_dump185 -p /var/lib/asterisk/astdb will show the complete database tree. Where can this db1_dump185 be found using Debian packages ? Regards ___ -- 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
