Re: [asterisk-users] CLI History
On Mon, Dec 11, 2006 at 10:31:41AM -0700, Douglas Garstang wrote: What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. Let's ignore the tone of the question, and try to answer it. Here is what I consider may be wrong: 1. The shell does not exit on Ctrl-D / EOF 2. When the shell does exist on Ctrl-C (SIGHUP) 3. When it exists on Ctrl-C (SIGHUP) it does not save history. 4. (Potentially a problem) saving stop now in the history. For starters, I hope you agree that (4) is not the real problem here. (1) and (2) make the Asterisk shell different from standard shells. In s standard shell Ctrl-C is supressed (so it will only affect programs you run from it, and not the shell). Ctrl-D and end of input cause the shell to exit. (3) seems to be the real thing that bothers Douglas. And as usual, Douglas manages to complain with the wrong tone and thus getting people to flame him rather than consider his argument seriously. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with IM
On Tue, Dec 12, 2006 at 09:55:52AM +0700, Mochamad Susantok wrote: How do i patch file chan_sip.so ? I use asterisk with Debian distro not asterisk-XXX.tar.gz You can still rebuild the package with the extra patch. Not a big problem. However: Mochamad Susantok wrote: Hi all, Howto configure asterisk 1.2.13 (debian-base) with support Instant Messaging, especially using client Xlite v.3. Im using my patched chan_sip.c for that. http://www.voiprakyat.or.id/download/server/asterisk/sip-messaging/1.2.13/ If I were to consider if to include this in Debian, the first thing I'd ask: is this code that is going to be merged into the main package? The README refers me to http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging where I see the following: Comment by oej: Although this is a great SIP patch, it will not be included in Asterisk due to it's one-channel perspective. As Juraj says, this is a hack and proof-of-concept. There are work going on to create a multi-protocol IM and presense solution for Asterisk. So could anybody erlaborate on this patch? Where exactly is it known to work? What problems is it known to have? What are its limitations? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] func_curl fails to compile, asterisk1.4
Hi. After successfully running ./configure I run make. When running make I get the following error: func_curl.c: In function `curl_internal': func_curl.c:95: `CURLOPT_NOSIGNAL' undeclared (first use in this function) func_curl.c:95: (Each undeclared identifier is reported only once func_curl.c:95: for each function it appears in.) make[1]: *** [func_curl.o] Error 1 make: *** [funcs] Error 2 This happens in the beta3 release as well as the HEAD revision in SVN. Has anybody else came across this problem? Please help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to open pseudo channel for timing... Sound may be choppy.
On Mon, Dec 11, 2006 at 04:18:47PM -0500, Phil Finkler wrote: Any idea what causes the warning Unable to open pseudo channel for timing... Sound may be choppy.? Any ideas what I need to resolve this? I do have the zaptel module installed but don't have a zaptel card. I'm guessing this has to do with ztdummy? I'm running Debian and installed asterisk, zaptel, and zaptel-source from the backports. Any information appreciated! modprobe ztdummy This hsould be run on boot (by the zaptel init.d script) if don't have a card. What kernel do you have? Have you built zaptel-modules? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail App
Hi All Is there an app or clever piece of dial plan that allows you to pull a call back from voicemail when you missed it? I am often on the phone and see I have another call, I hang up and the call has already gone to voicemail, is there anyway to pull this back to my phone? On our current BCM we use *XX and it pulls the call out of the voicemail system. Hope this makes sense. Thanks SP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: func_curl fails to compile, asterisk1.4
I got it sorted by myself in the meantime. I had version 7.9.8 of CurlLib installed. I upgraded to 7.16.0 and everything compiled just fine. Why didn't the configure script check for this version dependency of CurlLib? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400
On Mon, Dec 11, 2006 at 08:08:18PM -0500, Time Bandit wrote: [channels] context=default signalling=fxs_ls ;channel=1-16 usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 ;accountcode=lss0101 answeronpolarityswitch=yes hanguponpolarityswitch=yes To the best of my knowledge, all the settings you put after defining the channles (channel= line) are useless. You have to set all the settings BEFORE you define the channels. Should be. However in practice after the first reload all of them will be applied (in this specific case). /me points again to genzaptelconf that should have made this thread unnecessary. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Now OffTopic: VPN As SIP Tunneling?
Am Dienstag, den 12.12.2006, 01:32 -0600 schrieb Henry J. Cobb: Luki [EMAIL PROTECTED] wrote: You don't gain anything QOS-wise by going through a tunnel, except hiding your traffic in case your ISP purposefully assigns lower priority to VoIP traffic and doesn't do it to OpenVPN/GRE/insert your favorite tunnel protocol traffic. It's a pity that OpenVPN doesn't have an option to hide as https requests (and handle the double-TCP problem internally) or even better yet gif uploads over http. It very well has, if you think about getting through a HTTP proxy, faking a HTTPS session. It is not uncommon for student appartements internet to be restricted to HTTP and HTTPS only (and that, through a proxy, CONNECT restricted to those two ports). If you have OpenVPN running on port 443, TCP, you can use the proxy options to get a connection. Has been in use with the U of Edinburgh where a friend of mine needed a phoneline during his Erasmus term. Got a sipgate account for him, and tunneled asterisk VoIP through OpenVPN... worked like a charm. Using something like GIF-Upload/Download would require a much deeper lever of network stack, basically most of a webserver would have to be setup. It seems unlikely there are enough requests for this sort of functionality... Unless it is you who starts coding :) BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI History
Am Montag, den 11.12.2006, 22:19 -0800 schrieb Luki: thats prety smart... think hard.. wot was the command u gave to exit the CLI?? OK, come on everyone. This is getting ridiculous. That's the entire point that stop now was NOT the last command on the CLI, yet it shows up at the most recent upon recall with the Up key. I have the same, except in my case it's stuck on show channels (which is rather convenient so I didn't complain). And yes, it doesn't matter if I exit the CLI with Ctrl+C or exit. In my case it's probably a permission issue since I run * non-root and chroot'ed. Either way, I don't see why the history could not be save upon exit with Ctrl+C -- the mySQL client does it. Actually using quit the history is saved. That makes three more keystrokes, but using Ctrl+C to end something gracefully just makes me shudder. I even do not like using that combination for CopyPaste (preferring CtrlIns)... I think a point has been made that having stop now in the history is inconvenient for many people. My personal opinion is that NOT storing the commands - exit - stop - quit in the history would improve its usability. Getting Ctrl+D to work would help too. IIRC it boils down to catching SIGHUP (instead of/additional to SIGINT for Ctrl+C). I do not know wether those discussions have been had on the developers list, so it could be worth to bring it to attention over there. The code changes in question would expectedly be trivial; if there was some kind of consensus (more complicated) someone e.g. me could do that. (Following the old rule of OpenSource Feature Requests: If you want something done, best offer to do it yourself... :-) Best Regards, Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem in making outbound calls in PRI
Is there anybody who can help me out on this ? I am pretty much lost in forums and docs, and I m getting nowhere. - Danny Danny wrote: Hey everyone ! I have a problem in making outbound calls in PRI connection. I have E1 PRI airtel connection [ India ] [ asterisk-1.2.12.1 on CentOS 4.4 ] zaptel.conf -- [channels] language=en usecallerid = yes hidecallerid = no callwaiting=yes threewaycalling = yes usecallingpres=yes transfer = yes echocancel = yes echotraining = yes immediate = no ;group=0 ;context = from-pstn ;signalling = fxs_ks ;channel = 1 callwaitingcallerid=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes answeronpolarityswitch=yes rxgain=0.0 txgain=0.0 ; faxdetect=incoming ;- immediate=no overlapdial=yes pridialplan=national prilocaldialplan=national group=0 context = from-pstn callerid=asreceived switchtype = euroisdn signalling = pri_cpe channel = 1-15,17-31 ; --- ; zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 #fxsks=1 # Global data loadzone= us defaultzone = us ; --- When i try to make an outbound call, I get this in the error message : - -- Executing Dial(SIP/1001-0879af90, ZAP/g0/908239793) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/9821795097 -- Zap/1-1 is proceeding passing it to SIP/1001-0879af90 -- Channel 0/1, span 1 got hangup request -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) CLI I am not sure of what is wrong with my zaptel config. Any suggestions ? - Danny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using SIP with NAT (technical code question)
You might want to pass that one by the asterisk-dev list. Bob... On Mon, 2006-12-11 at 15:18 -0800, je . wrote: My mistake, I misread it. So if a hostname is provided (e.g. [EMAIL PROTECTED]) instead of an IP (e.g. 123.123.123.123) and the recipient of the INVITE is not using NAT then ast_gethostbyname will be run - is that correct? In this case, why the distinction between a NATted and non_NATted implementation? --- Bob Chiodini [EMAIL PROTECTED] wrote: It looks to me that if the test clause is false then ast_gethostbyname is called. Presumably not needed when NAT is enabled. Bob... je . wrote: In chan_sip.c, line 5876 (Asterisk-1.2.13), the function parse_ok_contact returns whether the host that requested an invite is a valid or invalid host. In line 5925 the following clause is tested: if (!(ast_test_flag(pvt, SIP_NAT) SIP_NAT_ROUTE)) hp = ast_gethostbyname(n, ahp); If this clause is true then Asterisk will attempt to retrieve the IP address by using the hostname provided in the invite. My question is, is this test always going to be true if a user (who receives the invite) uses NAT? (this is set up in sip.conf as nat=yes) Is there a reason why this was set up only for NAT? Thanks, Jez __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VPN As SIP Tunneling?
Am Montag, den 11.12.2006, 18:48 -0500 schrieb Barry Fawthrop: Hi Anselm Thanks for your input Yes I was thinking of using OpenVPN so it was good to hear your experiences I'm not so much concerned with the encryption of traffic etc.. But the Level of QoS. If my IP Phone set QoS and the VoIP Termination provider's * PBX sets QoS And we now connected via a VPN tunnel. We should be able to guarantee Quality due to the Tunnel. No, that is not true because you have no control over the tunnel packets. For an analogy, you can buy yourself first-class tickets for a transatlantic flight, but that will not help the plane you sit in to skip queue on the airport to take off earlier. You still have to rely on the underlying transport. The main issue is would I expect a higher latency ? Compared to non-VPN: Yes, latency is to be expected a little higher. It will probably not matter much though because it should be a magnitude smaller of the latency incurred by DSL links and the like. Someone (googled for numbers) claims the typical increase in roundtrip time to be less than 5 msec. and (2) If I were using a 1 Mbps connect would I have less bandwidth due to overheads. That where I could do 8 concurrent calls x 115 bps 920 kbps I could now only do 6 or will I still be able to do 8 ? I cannot say. I would expect a bandwidth overhead of 7% to 8%, from the numbers I saw on the web, so 7 streams could be OK, _possibly_ 8. You will have to try out. A question though is why you have 115kpbs/call/sec - that is quite significantly above ISDN call quality (64kbps/call/sec). You could always use less fat codecs... if your phones support those. alaw or ulaw should be supported by nearly all devices out there... and as uncompressed codecs are even below the numbers you gave. GSM restricts quality a lot, but you could nearly transport a GSM stream over an avian carriers link ( http://www.faqs.org/rfcs/rfc1149.html ) SCNR Best regards, Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem in making outbound calls in PRI
Danny wrote: Is there anybody who can help me out on this ? I am pretty much lost in forums and docs, and I m getting nowhere. Danny, You've got a lot of stuff in there that isn't used for a PRI/ISDN. Mine setup attached. This if for a T1, and in the US. Please make adjustments for your area: [zaptel.conf] span=1,1,0,esf,b8zs defaultzone=us loadzone=us bchan=1-23 dchan=24 span=2,0,0,esf,b8zs fxsks=25-32 fxoks=33-48 defaultzone=us loadzone=us [zapata.conf] [channels] musiconhold=tape switchtype=national context=pri signalling=pri_cpe group=1 echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=-1.0 txgain=-4.0 busydetect=no callprogress=no pridialplan=unknown usercallerid=yes callerid=asreceived channel = 1-23 -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Re: CLI History
But ctrl-c is 3 less keystrokes than exit\n ! LOL ;-) Maybe put a bug in bugs.digium.com asking for Ctrl-C to be caught and processed as exit. -- -- Steven http://www.glimasoutheast.org Douglas Garstang [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] But ctrl-c is 3 less keystrokes than exit\n ! -Original Message- From: Steven [mailto:[EMAIL PROTECTED] Sent: Monday, December 11, 2006 2:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: CLI History Don't hit Ctrl-C! If I type ? in the CLI, Ctrl-C is not listed as a command. *CLI ! abort add ael agent agi cdr databasedebug dnsmgr dontdump dundi extensions feature group helpiax2include indication init loadlocal logger meetme mgcp mixmonitor moh no pri realtimereload remove restart rtp set showsip skinny soft stopunload zap The funny thing is that neither is exit. Type exit when exiting asterisk CLI, and it will close out properly. -- -- Steven http://www.glimasoutheast.org Todd- Asterisk [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] short version: me too long version: The same thing happens on my asterisk boxes - both built with the latest trixbox image... perhaps that's a factor? My history is always restart now, although I typically connect and run sip show peers. I haven't typed restart now in a long time, but that is the first thing when I hit up-arrrow upon connecting I have had history written to when I type 'exit' at the console instead of ctrl-c. I haven't tested though as the school bus just arrived ;) Todd On Dec 11, 2006, at 3:35 PM, Douglas Garstang wrote: -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Monday, December 11, 2006 12:57 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CLI History On Mon, 2006-12-11 at 11:17 -0800, Carla Schroder wrote: On Monday 11 December 2006 9:31 am, Douglas Garstang wrote: What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI A No such command 'A' (type 'help' for help) hera*CLI B No such command 'B' (type 'help' for help) hera*CLI C No such command 'C' (type 'help' for help) hera*CLI D No such command 'D' (type 'help' for help) hera*CLI E No such command 'E' (type 'help' for help) hera*CLI [10:[EMAIL PROTECTED](pbx3):~]# asterisk -trv Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. == === Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI stop now -- I pressed the UP arrow upon re-entering the console! I'm a bit confused by your example. What are A,B,C, etc? To exit the Asterisk console, I type 'exit'. Asterisk continues to run, as it should. To re-enter the console I use asterisk -rvvv. He was demonstrating how the CLI history shows stop now as the last command (which um... it's a history? you're last command is gonna be the um... last command you ran... i.e. stop now). For crying out loud, why is this so hard to understand? It isn't rocket science. I said that when I exit the CLI and re-enter, no matter what my previous set of commands was, when I hit the UP arrow key, it was always 'stop now'. 'Stop now' WAS NOT MY PREVIOUS COMMAND. For the person that suggested maybe unknown commands are not added to the history... hera*CLI show channels Channel Location State Application(Data) 0 active channels 0 active calls hera*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 0 active SIP channels hera*CLI (I Pressed Ctrl-c here) [13:[EMAIL PROTECTED](pbx3):~]# asterisk -trv Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public
[asterisk-users] SPA2100 sends an unexpected BYE message when transmitting a FAX
Hi everyone, I'm trying to send a FAX with the following configuration: Analog FAX machine (OKI) -SPA21000-LAN-Asterisk PSTN I'm restricted to use passthru mode for faxing, instead of T.38 protocol, because the Asterisk box is running v1.2 and cannot be changed as it is in a heavy production environment. Anyway, it should work in passthru mode (G.711a) as the ATA and the Asterisk are in the same LAN with very low traffic. The problem arises when I try to send a fax: the Asterisk server initiates the call and, after a few seconds, the Linksys hangs the call by sending a BYE message: DEBUG[7416]: chan_sip.c:11375 handle_request: Received ACK (6) - Command in SIP ACK DEBUG[7416]: chan_sip.c:1396 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #258 DEBUG[7416]: chan_sip.c:1407 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 102: Match Found -- SIP read from 192.168.6.222:5060: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.6.222:5060;branch=z9hG4bK-6b0d80f5 From: 201 sip:[EMAIL PROTECTED];tag=c6bf15cfef0f4e11o0 To: sip:[EMAIL PROTECTED];tag=as715a2601 Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username=201,realm=asterisk,nonce=36741c67,uri=sip:[EMAIL PROTECTED],algorithm=MD5,response=94f0139b69bb01ddc4aa362ab3edc130 User-Agent: Linksys/SPA2100-3.3.6 I'm using the following features: - Network jitter buffer: very high - Jitter buffer adjustment: disable - Call Waiting: no - 3 Way Calling: no - Echo Canceller: no - Silence suppression: no - Preferred Codec: G711a - Use pref. codec only: yes - Silence Threshold = medium - Echo Canc Enable = no - Echo Canc Adapt Enable = no - Echo Supp Enable = no - FAX CED Detect Enable = no - FAX CNG Detect Enable = no - FAX Passthru Codec = G711a - FAX Passthru Method = NSE - FAX Process NSE = yes - FAX Disable ECAN = no - FAX Codec Symmetric = yes - DTMF Tx Method = auto - Hook Flash Tx Method = none - Release Unused Codec = yes I have checked the SPA2100's logs, but I can't see anything of interest (and I couldn't find any documentation about this logs at Sipura's website). Has anyone suceed in sending a fax in a scenario like this? I would appreciate any help on this point. Best regards, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 to SIP protocol translation overhead?
One main disadvantage would be the media stream will pass through asterisk ( no reinvites like sip-sip ) but its not a problem if client pc'a and your asterisk server are on same network .Sip-iax conversion takes less cpu but it will be more if codec transcoding is involved . On 12/12/06, David Thomas [EMAIL PROTECTED] wrote: Just wondering if there is much CPU overhead in the translation from IAX2 to SIP, and how taxing this function is as compared to transcoding. We're trying to build an efficient system and would like to avoid taxing the CPU as much as possible. Our upstream service provider is 100% SIP, however we'd like to use IAX2 in our network as well, if it does not cause too much overhead. Not sure if it matters, but we will be running aprox 100 simultaneous calls. Thanks, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup Party
Hello, Is there a way to find out which party hanged up the call. Generally this is reported as Local disconnet/Remote disconnect in callcenter environments. Thanks. Idris Information and Communication Technologies Manager Vodatech ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup Party
On Tue, 12 Dec 2006 15:27:06 +0200 Idris AVCI [EMAIL PROTECTED] wrote: Hello, Is there a way to find out which party hanged up the call. Generally this is reported as Local disconnet/Remote disconnect in callcenter environments. This is already written to the queue_log e.g. 1165572107|1165572085.354|french|Local/[EMAIL PROTECTED]|COMPLETEAGENT|20|2 or 1165495361|1165495218.23|french|SIP/1337-08234748|COMPLETECALLER|6|137 gdh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hangup Party
Thanks Gavin. We are not using built-in acd functions. Is there any way to report this in dialplan functions ? -Original Message- From: Gavin Hamill [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 12, 2006 3:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hangup Party On Tue, 12 Dec 2006 15:27:06 +0200 Idris AVCI [EMAIL PROTECTED] wrote: Hello, Is there a way to find out which party hanged up the call. Generally this is reported as Local disconnet/Remote disconnect in callcenter environments. This is already written to the queue_log e.g. 1165572107|1165572085.354|french|Local/[EMAIL PROTECTED]|COMPLETEAGENT|20|2 or 1165495361|1165495218.23|french|SIP/1337-08234748|COMPLETECALLER|6|137 gdh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Unable to open pseudo channel for timing...Sound may be choppy.
Thanks for the help. I added zaptel and ztdummy to startup and the warning seems to have gone away! Phil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, December 12, 2006 4:01 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Unable to open pseudo channel for timing...Sound may be choppy. On Mon, Dec 11, 2006 at 04:18:47PM -0500, Phil Finkler wrote: Any idea what causes the warning Unable to open pseudo channel for timing... Sound may be choppy.? Any ideas what I need to resolve this? I do have the zaptel module installed but don't have a zaptel card. I'm guessing this has to do with ztdummy? I'm running Debian and installed asterisk, zaptel, and zaptel-source from the backports. Any information appreciated! modprobe ztdummy This hsould be run on boot (by the zaptel init.d script) if don't have a card. What kernel do you have? Have you built zaptel-modules? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk from Debian Packages
Alex, Thanks for the help. I've installed Asterisk and Zaptel from the backports and so far so good! Phil From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Sent: Monday, December 11, 2006 11:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk from Debian Packages You can run Asterisk 1.2 in sarge using the packages in backports. Just add: deb http://www.backports.org/debian/ sarge-backports main contrib non-free to /etc/apt/sources.list then apt-get update and then apt-get -t sarge-backports install asterisk (you can also pin-priority asterisk's packages, look at APT documentation). -Alex On 12/10/06, Phil Finkler [EMAIL PROTECTED] wrote: Hi all, I've gotten asterisk installed on Debian only to realize that the packaged version is 1.0.7. Is there a reason why they're not up to a 1.2.x release? I'm building a system for production and I'm wondering if I should remain at this old version or if there are any serious issues with 1.2.13 on Debian? Should I be able to do an apt-get from unstable and get 1.2.13 and be on my happy way? Thanks for the help on a stupid question, Phil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Loosing IAX connection between offices
I converted my connections from IAX to SIP and still having the same problem. I'm loosing connection between B and A. There is also an Office C. My problems seem to be with Office B. Now that I have switched to SIP, I still have the same problem when the IP address changes at Office C, Office B looses connection and can not be restablished with a SIP RELOAD. I think the problem is DNS related, either in Asterisk or the router. Though, this is getting above my head at this point. Can anyone point me in right direction? On 12/4/06, Louis-David Mitterrand [EMAIL PROTECTED] wrote: On Thu, Nov 30, 2006 at 08:52:50AM -0600, DM wrote: Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Short story: IAX is still crap in 1.2.13 (haven't tested 1.4), it's unreliable and perfectly good hosts will become UNREACHABLE for no apparent reason, while SIP connections keep going through. For trunking, avoid IAX and use SIP. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] repost gain problem with asterisk and zaptel 1.4
Hey guys, I'm having some trouble with gain settings using a Wildcard X100P and Zaptel 1.4 Beta 2 on Asterisk 1.4 beta 3. My transmit volume between an IAX client (idefisk) and a POTS device through my 1.4 box is great, but my receive volume is terrible. I can hardly hear a word they say. I've cranked my gains up to 100, inserted the wcfxo module with boost=1, but haven't had any luck figuring this out. Using ztmonitor, the best I can get on my RX side is 3 #'s (###)) and that's if I really talk loudly into the phone. Any pointers on other places I can look? Volume is great on IAX to IAX, only poor on calls from IAX to POTS. I haven't tried the other direction because this box isn't setup to receive POTS calls. -- Jason The place where you made your stand never mattered, only that you were there... and still on your feet ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outgoing call on ISDN PRI
HEllo list ! When user A calls user B via Asterisk (Users A and B are registered on the same Asterisk server ) and an ISDN PRI, user B phone always shows Asterisk server telephone number. How to hide it and how to forward user A number ? We tried usecallerid, callerid, hidecallerid, restrictcid, usecallingpres in zapata.conf but we always see Asterisk server telephone number ! Thanks you! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add include statement into Realtime static
you must use the switch command. I am not sure, but I think you should configure config realtime also, otherwise this command will be in extensions.conf Take a look in voip-info.org 2006/12/12, Tielin Xu [EMAIL PROTECTED]: Hi List: I can not find out an example how to store include = context name statement into Realtime static. Please help me on this one. Thanks, Tielin ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP and IAX configuration from LDAP
Hi All, Had anyone got an idea of there exists an LDAP backend for SIP and IAX? I've read that there is a patch for LDAP realtime, but I hadn't seen any type of relevant configuration information. Any information on the above would be highly appreciated. Regards, Nir S ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone using Ranch Networks products for Load Balancing in a SIP environment?
Looking for info recommendations for SIP load balancing, thanks in advance! Cory Andrews ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] repost gain problem with asterisk and zaptel 1.4
On Tue, Dec 12, 2006 at 08:57:05AM -0600, jason wrote: Hey guys, I'm having some trouble with gain settings using a Wildcard X100P and Zaptel 1.4 Beta 2 on Asterisk 1.4 beta 3. My transmit volume between an IAX client (idefisk) and a POTS device through my 1.4 box is great, but my receive volume is terrible. I can hardly hear a word they say. I've cranked my gains up to 100, inserted the wcfxo module with boost=1, but haven't had any luck figuring this out. Using ztmonitor, the best I can get on my RX side is 3 #'s (###)) and that's if I really talk loudly into the phone. Any pointers on other places I can look? Volume is great on IAX to IAX, only poor on calls from IAX to POTS. I haven't tried the other direction because this box isn't setup to receive POTS calls. Could you post your zapata.conf ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to add include statement into Realtime static
The 'include =' statement works fine for us in realtime static. Doug. -Original Message- From: Fran Oliveira [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 12, 2006 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [asterisk-users] How to add include statement into Realtime static you must use the switch command. I am not sure, but I think you should configure config realtime also, otherwise this command will be in extensions.conf Take a look in voip-info.org http://voip-info.org/ 2006/12/12, Tielin Xu [EMAIL PROTECTED]: Hi List: I can not find out an example how to store include = context name statement into Realtime static. Please help me on this one. Thanks, Tielin ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Input on Dundi
Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock http://www.bochterservices.com/?t=TFdid For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI problema
Hi all. I've written a AGI in C language. It
receive the asterisk variables to identify the caller. After, it dial
to destination. When caller or the called hangup the phone, asterisk
returns me '200 result=-1'. For this, asterisk never execute next step,
priority 2. This is very important to me, because priority 2 do the
billing. Below I give you the debug message:
-- Executing agi("SIP/provale-7473", "dialer|551236337388")
-- Launched AGI Script /usr/local/share/asterisk/agi-bin/dialer
AGI Tx agi_request: dialer
AGI Tx agi_channel: SIP/provale-7473
AGI Tx agi_language: br
AGI Tx agi_type: SIP
AGI Tx agi_uniqueid: 1165939032.131
AGI Tx agi_callerid: provale
AGI Tx agi_calleridname: Provale
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callington: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: 01236337388
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: default
AGI Tx agi_extension: 01236337388
AGI Tx agi_priority: 1
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx
AGI Rx exec dial "sip/go2call/551236337388|60|TtS(3600)"
-- AGI Script Executing Application: (dial) Options:
(sip/go2call/551236337388|60|TtS(3600))
-- Setting call duration limit to 3600 seconds.
-- Called go2call/551236337388
-- SIP/go2call-3fd0 is making progress passing it to
SIP/provale-7473
-- SIP/go2call-3fd0 answered SIP/provale-7473
-- Attempting native bridge of SIP/provale-7473 and SIP/go2call-3fd0
AGI Tx 200 result=-1
-- AGI Script dialer completed, returning 0
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Re: [asterisk-users] Input on Dundi
I use it to handle calls between multiple sites connected over a wan. It works great, I finally understood the concepts after the Astricon presentation on clustering with dundi. On 12/12/06, Al Bochter [EMAIL PROTECTED] wrote: Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock http://www.bochterservices.com/?t=TFdid For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager
Hello, I am not an asterisk expert but i am developing a web application that is using asterisk. I would like to know if it is possible to configure a Manager to only monitor a special extension, and of course how to do that. The application is written in java and is using asterisk-java. Right now i have one manager that i am connected to and i receive all the events but i would like to have some kind of administrator and user. The administrator manager can receive all events but the normal user (agent) should only receive the events that are associated to its extension. I would have some kind of user 1010 (the actual extension and username too) Let's say that in manager.conf i would have again some user 1010 but i would like that this user can only see the events associated to the extension 1010 ... Does it makes any sens, and how to do that? Regards, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Input on Dundi
It's just a shame there isn't complete documentation available. -Original Message- From: Bruce Reeves [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 12, 2006 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Input on Dundi I use it to handle calls between multiple sites connected over a wan. It works great, I finally understood the concepts after the Astricon presentation on clustering with dundi. On 12/12/06, Al Bochter [EMAIL PROTECTED] wrote: Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock http://www.bochterservices.com/?t=TFdid For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBX http://www.bochterservices.com/?j=PBXt=email t=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=gold http://www.bochterservices.com/?j=goldt=email t=email For new and used security items http://www.bochterservices.com/?j=store http://www.bochterservices.com/?j=storet=email_security t=email_security ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Manager
CLIPPED I would have some kind of user 1010 (the actual extension and username too) Let's say that in manager.conf i would have again some user 1010 but i would like that this user can only see the events associated to the extension 1010 ... CLIPPED I am pretty sure that using the proxy, astmanproxy, you can achieve this goal. It is recommended to use the proxy so that there is only one connection to the server and all the other applications will connect to the proxy. -Jonathan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Input on Dundi
Douglas. I can't agree more. Thats VoIP things for you little to no documentation :-( Well thats ok, I am working on some documentation for Asterisk and other Distros. a2billing is one I am working on dundi will be next. And others I will post the links when its ready and right. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Douglas Garstang wrote: It's just a shame there isn't complete documentation available. -Original Message- *From:* Bruce Reeves [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, December 12, 2006 9:07 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Input on Dundi I use it to handle calls between multiple sites connected over a wan. It works great, I finally understood the concepts after the Astricon presentation on clustering with dundi. On 12/12/06, *Al Bochter* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock http://www.bochterservices.com/?t=TFdid For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email http://www.bochterservices.com/?j=PBXt=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security http://www.bochterservices.com/?j=storet=email_security ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0657-0, 12/12/2006 - 12/12/2006 11:34:57 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI problema
There is something which I could do to execute priority 2? It's
possible my agi have programming error?
Eduardo wrote:
Hi all. I've written a AGI in C language. It
receive the asterisk variables to identify the caller. After, it dial
to destination. When caller or the called hangup the phone, asterisk
returns me '200 result=-1'. For this, asterisk never execute next step,
priority 2. This is very important to me, because priority 2 do the
billing. Below I give you the debug message:
-- Executing agi("SIP/provale-7473", "dialer|551236337388")
-- Launched AGI Script /usr/local/share/asterisk/agi-bin/dialer
AGI Tx agi_request: dialer
AGI Tx agi_channel: SIP/provale-7473
AGI Tx agi_language: br
AGI Tx agi_type: SIP
AGI Tx agi_uniqueid: 1165939032.131
AGI Tx agi_callerid: provale
AGI Tx agi_calleridname: Provale
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callington: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: 01236337388
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: default
AGI Tx agi_extension: 01236337388
AGI Tx agi_priority: 1
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx
AGI Rx exec dial "sip/go2call/551236337388|60|TtS(3600)"
-- AGI Script Executing Application: (dial) Options:
(sip/go2call/551236337388|60|TtS(3600))
-- Setting call duration limit to 3600 seconds.
-- Called go2call/551236337388
-- SIP/go2call-3fd0 is making progress passing it to
SIP/provale-7473
-- SIP/go2call-3fd0 answered SIP/provale-7473
-- Attempting native bridge of SIP/provale-7473 and SIP/go2call-3fd0
AGI Tx 200 result=-1
-- AGI Script dialer completed, returning 0
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Re: [asterisk-users] repost gain problem with asterisk and zaptel 1.4
set the rxgain= option BEFORE the channel= lines! jason wrote: Hey guys, I'm having some trouble with gain settings using a Wildcard X100P and Zaptel 1.4 Beta 2 on Asterisk 1.4 beta 3. My transmit volume between an IAX client (idefisk) and a POTS device through my 1.4 box is great, but my receive volume is terrible. I can hardly hear a word they say. I've cranked my gains up to 100, inserted the wcfxo module with boost=1, but haven't had any luck figuring this out. Using ztmonitor, the best I can get on my RX side is 3 #'s (###)) and that's if I really talk loudly into the phone. Any pointers on other places I can look? Volume is great on IAX to IAX, only poor on calls from IAX to POTS. I haven't tried the other direction because this box isn't setup to receive POTS calls. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto answer when already on a call
I have a customer that has Aastra phones on an Asterisk 1.2.13 system. The big boss want to be able to interrupt someones phone even if they are in the middle of a call. What he wants is basically that when he dials the busy extension that he gets on the speaker so he can say something to that person. I tried to use the example from the paging section of the Wiki and if there is no other call on the phone then I can get directly on the speaker. But if that phone already has another call then it gives me a busy tone. The phone can handle at least 3 calls (Aastra 9133i) and if I make a regular call I do get into the second line. Is there a way to make the phone auto answer on speaker and interrupting the first call? -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chà vez Prats Director de Tecnologìa +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] repost gain problem with asterisk and zaptel 1.4
Most certainly. This is my current file. I've toyed with the gain settings, going as high as 100 on rxgain and even toyed with txgain for a bit. It's very bare because I haven't invested any further time into this machine until I get the gain figured out. On a side note, I've used this card in a Xen VM running 1.2 and did not have any gain issues. -- ; Zapata telephony interface [trunkgroups] ; [channels] ; context=incoming switchtype=national ; signalling=fxs_ks channel=1 usecallerid=yes cidsignalling=bell hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default Tzafrir Cohen wrote: On Tue, Dec 12, 2006 at 08:57:05AM -0600, jason wrote: Hey guys, I'm having some trouble with gain settings using a Wildcard X100P and Zaptel 1.4 Beta 2 on Asterisk 1.4 beta 3. My transmit volume between an IAX client (idefisk) and a POTS device through my 1.4 box is great, but my receive volume is terrible. I can hardly hear a word they say. I've cranked my gains up to 100, inserted the wcfxo module with boost=1, but haven't had any luck figuring this out. Using ztmonitor, the best I can get on my RX side is 3 #'s (###)) and that's if I really talk loudly into the phone. Any pointers on other places I can look? Volume is great on IAX to IAX, only poor on calls from IAX to POTS. I haven't tried the other direction because this box isn't setup to receive POTS calls. Could you post your zapata.conf ? -- Jason The place where you made your stand never mattered, only that you were there... and still on your feet ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Input on Dundi
I guess that depends on what you mean, I got a good overview in the Asterisk bootcamp from Lief, but then the whitepaper and presentation by JR Richardson sealed the deal on our use of dundi to dynamicly route calls. On 12/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: It's just a shame there isn't complete documentation available. -Original Message- *From:* Bruce Reeves [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, December 12, 2006 9:07 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Input on Dundi I use it to handle calls between multiple sites connected over a wan. It works great, I finally understood the concepts after the Astricon presentation on clustering with dundi. On 12/12/06, Al Bochter [EMAIL PROTECTED] wrote: Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock http://www.bochterservices.com/?t=TFdid For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP and IAX configuration from LDAP
there is a sample configuartion file that you can start working with in the res_config_ldap group branch. http://svn.digium.com/view/asterisk/team/group/res_config_ldap/configs/res_ldap.conf.sample?view=markup -anthony - Original Message - From: Nir Simionovich [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, December 12, 2006 9:20:29 AM GMT-0600 US/Central Subject: [asterisk-users] SIP and IAX configuration from LDAP Hi All, Had anyone got an idea of there exists an LDAP backend for SIP and IAX? I’ve read that there is a patch for LDAP realtime, but I hadn’t seen any type of relevant configuration information. Any information on the above would be highly appreciated. Regards, Nir S ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] long busy()
hi list,
I set up a new asterisk machine with asterisk 1.2.13 and misdn 0.3.1rc27.
I use an e1 card with sip clients. My extensions look like this:
[E1]
snip...snip
exten = 33006733,1,Set(CALLED=${EXTEN})
exten = 33006733,2,Dial(SIP/[EMAIL PROTECTED])
exten = 33006733-ANSWER,3,Answer()
[SIP]
exten = _X.,1,Noop()
exten = _X.,2,SetCallerPres(allowed_passed_screen)
exten = _X.,3,Dial(mISDN/g:E1/${EXTEN},40)
exten = _X.-BUSY,4,Busy(1)
But whenever a sip client calls to an exten that is busy through e1 I get busy
tones for 10s before I get disconnected. But I want to have it only for 1s.
Does anyone know how to fix that?
regards, Christophorus
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Re: [asterisk-users] Anyone using Ranch Networks products for Load Balancing in a SIP environment?
We have a few clients running large CCs who are using them and seem quite happy with them. l. On Tue, 12 Dec 2006 16:21:36 +0100, Cory Andrews [EMAIL PROTECTED] wrote: Looking for info recommendations for SIP load balancing, thanks in advance! Cory Andrews -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI problema
On 14:43, Tue 12 Dec 06, Eduardo wrote: There is something which I could do to execute priority 2? It's possible my agi have programming error? You can use the h extension to do stuff after a channel is hungup. This is mostly used for billing or processing of the recording file of a call. If you really want to do it in priority 2 you can use the g option in the dial command. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI problema
I've never tried to dial from an AGI. Would setting a channel variable with the dial string and then dialing in the dial plan work for you? Then you could handle the hangup in the h extension and call your billing using deadagi(). On Tue, 12 Dec 2006, Eduardo wrote: There is something which I could do to execute priority 2? It's possible my agi have programming error? Eduardo wrote: Hi all. I've written a AGI in C language. It receive the asterisk variables to identify the caller. After, it dial to destination. When caller or the called hangup the phone, asterisk returns me '200 result=-1'. For this, asterisk never execute next step, priority 2. This is very important to me, because priority 2 do the billing. Below I give you the debug message: -- Executing agi(SIP/provale-7473, dialer|551236337388) -- Launched AGI Script /usr/local/share/asterisk/agi-bin/dialer AGI Tx agi_request: dialer AGI Tx agi_channel: SIP/provale-7473 AGI Tx agi_language: br AGI Tx agi_type: SIP AGI Tx agi_uniqueid: 1165939032.131 AGI Tx agi_callerid: provale AGI Tx agi_calleridname: Provale AGI Tx agi_callingpres: 0 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 0 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: 01236337388 AGI Tx agi_rdnis: unknown AGI Tx agi_context: default AGI Tx agi_extension: 01236337388 AGI Tx agi_priority: 1 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: AGI Tx AGI Rx exec dial sip/go2call/551236337388|60|TtS(3600) -- AGI Script Executing Application: (dial) Options: (sip/go2call/551236337388|60|TtS(3600)) -- Setting call duration limit to 3600 seconds. -- Called go2call/551236337388 -- SIP/go2call-3fd0 is making progress passing it to SIP/provale-7473 -- SIP/go2call-3fd0 answered SIP/provale-7473 -- Attempting native bridge of SIP/provale-7473 and SIP/go2call-3fd0 AGI Tx 200 result=-1 -- AGI Script dialer completed, returning 0 __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zapata.conf: cannot set txgain lower than -6.3 ?
Greetings everyone, I have a Digium TDM400P card with both an FXO and FXS module to connect to the phone company and to a standard phone. The problem is that the volume of my voice is going out too loud. I tried lowering the txgain value in zapata.conf to compensate, but all audio drops out completely if I set txgain to -6.4 or lower. If I set it to -6.3, then everything works (but still too loud). Is there a limitation as to how low txgain can be set? From voip-info.org, I was under the impression that the range was -100 to 100. Even the page at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf shows an example txgain of -15.9 Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long busy()
Christophorus Laube schrieb:
hi list,
I set up a new asterisk machine with asterisk 1.2.13 and misdn 0.3.1rc27.
I use an e1 card with sip clients. My extensions look like this:
[E1]
snip...snip
exten = 33006733,1,Set(CALLED=${EXTEN})
exten = 33006733,2,Dial(SIP/[EMAIL PROTECTED])
exten = 33006733-ANSWER,3,Answer()
[SIP]
exten = _X.,1,Noop()
exten = _X.,2,SetCallerPres(allowed_passed_screen)
exten = _X.,3,Dial(mISDN/g:E1/${EXTEN},40)
exten = _X.-BUSY,4,Busy(1)
But whenever a sip client calls to an exten that is busy through e1 I get busy
tones for 10s before I get disconnected. But I want to have it only for 1s.
Does anyone know how to fix that?
regards, Christophorus
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AFAIK the BUSY() command has nothing to do with the busy indication. You
can't pass anything to this command.
Check: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Busy
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Re: [asterisk-users] Asterisk Manager
Hello Jonathan, thank you for answering ... I read about astmanproxy but it cannot help me. I am using asterisk-java all my application is written in java too. I already have a kind of proxy ad I am not doing several connection to the asterisk manager. I am afraid this is not helping me much. Anyway, I have done this in my proxy but i thought i could avoid things like that in my code... I did not test the asterisk manager contexts and dial plan, so I wonder if I make a call via astman from 1010 to a GSM and that 1010 is in a context that is not allowing calls to GSM would astman execute it anyway or would it look also in the 1010 context? I am asking that because my system guys are not available until friday ... Jonathan k. Creasy wrote: CLIPPED I would have some kind of user 1010 (the actual extension and username too) Let's say that in manager.conf i would have again some user 1010 but i would like that this user can only see the events associated to the extension 1010 ... CLIPPED I am pretty sure that using the proxy, astmanproxy, you can achieve this goal. It is recommended to use the proxy so that there is only one connection to the server and all the other applications will connect to the proxy. -Jonathan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] repost gain problem with asterisk and zaptel 1.4
You MUST set the options BEFORE the channel= line or they won't be used. jason wrote: Most certainly. This is my current file. I've toyed with the gain settings, going as high as 100 on rxgain and even toyed with txgain for a bit. It's very bare because I haven't invested any further time into this machine until I get the gain figured out. On a side note, I've used this card in a Xen VM running 1.2 and did not have any gain issues. -- ; Zapata telephony interface [trunkgroups] ; [channels] ; context=incoming switchtype=national ; signalling=fxs_ks channel=1 usecallerid=yes cidsignalling=bell hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default Tzafrir Cohen wrote: On Tue, Dec 12, 2006 at 08:57:05AM -0600, jason wrote: Hey guys, I'm having some trouble with gain settings using a Wildcard X100P and Zaptel 1.4 Beta 2 on Asterisk 1.4 beta 3. My transmit volume between an IAX client (idefisk) and a POTS device through my 1.4 box is great, but my receive volume is terrible. I can hardly hear a word they say. I've cranked my gains up to 100, inserted the wcfxo module with boost=1, but haven't had any luck figuring this out. Using ztmonitor, the best I can get on my RX side is 3 #'s (###)) and that's if I really talk loudly into the phone. Any pointers on other places I can look? Volume is great on IAX to IAX, only poor on calls from IAX to POTS. I haven't tried the other direction because this box isn't setup to receive POTS calls. Could you post your zapata.conf ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Manager
Not meaning to argue with you but the proxy replaces the manager interface so it could most likely be a seamless replacement to your application. It was for all but one of my applications and the problem there was in the way I parsed the startup string. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Daniel Gradecak Sent: Tuesday, December 12, 2006 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Manager Hello Jonathan, thank you for answering ... I read about astmanproxy but it cannot help me. I am using asterisk-java all my application is written in java too. I already have a kind of proxy ad I am not doing several connection to the asterisk manager. I am afraid this is not helping me much. Anyway, I have done this in my proxy but i thought i could avoid things like that in my code... I did not test the asterisk manager contexts and dial plan, so I wonder if I make a call via astman from 1010 to a GSM and that 1010 is in a context that is not allowing calls to GSM would astman execute it anyway or would it look also in the 1010 context? I am asking that because my system guys are not available until friday ... Jonathan k. Creasy wrote: CLIPPED I would have some kind of user 1010 (the actual extension and username too) Let's say that in manager.conf i would have again some user 1010 but i would like that this user can only see the events associated to the extension 1010 ... CLIPPED I am pretty sure that using the proxy, astmanproxy, you can achieve this goal. It is recommended to use the proxy so that there is only one connection to the server and all the other applications will connect to the proxy. -Jonathan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Input on Dundi
On 12/12/06, Al Bochter [EMAIL PROTECTED] wrote: Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? We have been playing with DUNDi in a configuration similar to JR's whitepaper. Everything seems to be working fine but we have encountered a couple hurdles. Maybe others on the list have encountered these as well. 1.) When a registration server fails there doesn't seem to be an easy way to have clients automatically register to a new server. (our clients are mostly other asterisk boxes.) To solve this we are considering using DNS failover. 2.) If you plan to do any direct routing using the fullcontact address like what is shown in JR's whitepaper, you may find that fullcontact sometimes contains private network addresses. This makes it impossible to route inbound calls directly to the client. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip help for newbie
Does anyone know of any good step by step tutorials on getting sip set up? I have asterisk installed but can't seem to figure out how to get an account set up and connect from my xTen phone so I can try the demo. The tutorials I read online seem to go into voicepulse stuff and all and I don't have an account there so am a bit lost. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem in making outbound calls in PRI
On 12 Dec 2006, at 12:16, Doug Lytle wrote: Danny wrote: Is there anybody who can help me out on this ? I am pretty much lost in forums and docs, and I m getting nowhere. Danny, You've got a lot of stuff in there that isn't used for a PRI/ISDN. Mine setup attached. This if for a T1, and in the US. Please make adjustments for your area: Here is mine - for E1 EuroISDN (Q931 in the UK) #zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone = uk defaultzone=uk #zapata.conf [channels] language=en context=ntl switchtype=euroisdn pridialplan=local signalling=pri_cpe usecallerid=yes hidecallerid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes group=1 callgroup=1 pickupgroup=1 channel = 1-15,17-31 #- If that doesn't help, please send us the output of cat /proc/zaptel/1 And enable pri debugging in asterisk and send the output when you make a call. Don't be afraid to ask the provider what they are seeing... First time I put in a PRI I spent 2 days messing with it before I rang them, to be told that they hadn't enabled outbound calls yet! T. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem in making outbound calls in PRI
On 12 Dec 2006, at 12:16, Doug Lytle wrote: Danny wrote: Is there anybody who can help me out on this ? I am pretty much lost in forums and docs, and I m getting nowhere. Danny, You've got a lot of stuff in there that isn't used for a PRI/ISDN. Mine setup attached. This if for a T1, and in the US. Please make adjustments for your area: Here is mine - for E1 EuroISDN (Q931 in the UK) #zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone = uk defaultzone=uk #zapata.conf [channels] language=en context=ntl switchtype=euroisdn pridialplan=local signalling=pri_cpe usecallerid=yes hidecallerid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes group=1 callgroup=1 pickupgroup=1 channel = 1-15,17-31 #- If that doesn't help, please send us the output of cat /proc/zaptel/1 And enable pri debugging in asterisk and send the output when you make a call. Don't be afraid to ask the provider what they are seeing... First time I put in a PRI I spent 2 days messing with it before I rang them, to be told that they hadn't enabled outbound calls yet! T. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing call on ISDN PRI
On 12 Dec 2006, at 15:11, Michel wrote: HEllo list ! When user A calls user B via Asterisk (Users A and B are registered on the same Asterisk server ) and an ISDN PRI, user B phone always shows Asterisk server telephone number. How to hide it and how to forward user A number ? We tried usecallerid, callerid, hidecallerid, restrictcid, usecallingpres in zapata.conf but we always see Asterisk server telephone number ! I'm not getting a clear picture of how the ISDN PRI gets into it if both users are registered (SIP I assume) to the same asterisk. If the call actually goes out via a Public ISDN line, you have to get the provider to agree to let you set the outgoing number. Normally they will only let you set it to one of the inbound numbers that you have bought from them :-) If that doesn't help, please re-phrase the question... Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager
On 12 Dec 2006, at 16:27, Daniel Gradecak wrote: Hello, I am not an asterisk expert but i am developing a web application that is using asterisk. I would like to know if it is possible to configure a Manager to only monitor a special extension, and of course how to do that. The application is written in java and is using asterisk-java. Right now i have one manager that i am connected to and i receive all the events but i would like to have some kind of administrator and user. The administrator manager can receive all events but the normal user (agent) should only receive the events that are associated to its extension. I would have some kind of user 1010 (the actual extension and username too) Let's say that in manager.conf i would have again some user 1010 but i would like that this user can only see the events associated to the extension 1010 ... Does it makes any sens, and how to do that? The manager doesn't have any filters - per-se. You would need to add a layer in your asterisk-java program that filtered the channels/extensions you were interested in. The easiest thing might be to have your manager layer put the events into a lightweight (in memory?) database, then use some standard JDBC/servlets (or whatever) to query those events using the channel current user's as a key. Now, depending on what you are trying to do, there may be other ways to get there Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe Conferencing and Marked Mode
I am trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zapata.conf: cannot set txgain lower than -6.3 ?
Hi Steve, I tried txgain as low as -18 without any problem, but I never tried anything with decimal points. Andy On 12/12/06, Steve Hsieh [EMAIL PROTECTED] wrote: Greetings everyone, I have a Digium TDM400P card with both an FXO and FXS module to connect to the phone company and to a standard phone. The problem is that the volume of my voice is going out too loud. I tried lowering the txgain value in zapata.conf to compensate, but all audio drops out completely if I set txgain to -6.4 or lower. If I set it to -6.3, then everything works (but still too loud). Is there a limitation as to how low txgain can be set? From voip-info.org, I was under the impression that the range was -100 to 100. Even the page at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf shows an example txgain of -15.9 Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] repost gain problem with asterisk and zaptel 1.4
doh. thanks guys. It's always the simple stuff :-) Moved the channel statement around and works great. Eric ManxPower Wieling wrote: You MUST set the options BEFORE the channel= line or they won't be used. jason wrote: Most certainly. This is my current file. I've toyed with the gain settings, going as high as 100 on rxgain and even toyed with txgain for a bit. It's very bare because I haven't invested any further time into this machine until I get the gain figured out. On a side note, I've used this card in a Xen VM running 1.2 and did not have any gain issues. -- ; Zapata telephony interface [trunkgroups] ; [channels] ; context=incoming switchtype=national ; signalling=fxs_ks channel=1 usecallerid=yes cidsignalling=bell hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default Tzafrir Cohen wrote: On Tue, Dec 12, 2006 at 08:57:05AM -0600, jason wrote: Hey guys, I'm having some trouble with gain settings using a Wildcard X100P and Zaptel 1.4 Beta 2 on Asterisk 1.4 beta 3. My transmit volume between an IAX client (idefisk) and a POTS device through my 1.4 box is great, but my receive volume is terrible. I can hardly hear a word they say. I've cranked my gains up to 100, inserted the wcfxo module with boost=1, but haven't had any luck figuring this out. Using ztmonitor, the best I can get on my RX side is 3 #'s (###)) and that's if I really talk loudly into the phone. Any pointers on other places I can look? Volume is great on IAX to IAX, only poor on calls from IAX to POTS. I haven't tried the other direction because this box isn't setup to receive POTS calls. Could you post your zapata.conf ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jason The place where you made your stand never mattered, only that you were there... and still on your feet ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Input on Dundi
-Original Message- From: David Thomas [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 12, 2006 11:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Input on Dundi On 12/12/06, Al Bochter [EMAIL PROTECTED] wrote: Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? We have been playing with DUNDi in a configuration similar to JR's whitepaper. Everything seems to be working fine but we have encountered a couple hurdles. Maybe others on the list have encountered these as well. 1.) When a registration server fails there doesn't seem to be an easy way to have clients automatically register to a new server. (our clients are mostly other asterisk boxes.) To solve this we are considering using DNS failover. Wow. I remember when I raised this as an issue I was accused of being a Asterisk heretic. The solution suggested was to, increased load not-withstanding, bring your phone registration period right down. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 + New Firmware (8.2)
Hi All, Found out Cisco has some newer firmware available for the 7970 series of phones. New sip images are at version level 8.2 (instead of 8.0.2,8.0.3,8.0.4), posted Dec 10, 2006. This major jump in version numbers has fixed a few bugs (time zone not updating properly), but hasn't figured what some would consider to be showstoppers (registration not fully working, and mwi still not working). Just thought I would let you all know there's new firmware to mess around with! Also, to note, to get the phone to actually take this upgrade, and you're running your tftp server on a linux box, then you will need to rename one of the files for it to find it properly. # cd tftpdroot # mv jar70sip.8-2-0-55.sbn Jar70sip.8-2-0-55.sbn Calls in and Out work, though the phone still shows that dreaded red x next to the extension saying it's not registered. MWI is also still not working with 3 or 1 in the MWI indicator slot in the .xml file. And no, I won't email you the firmware, you need a cisco login to get one, so get a friend, or join cisco yourself! :) Happy Testing! Matt G http://www.voipphreak.ca ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Input on Dundi
On 12/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: David Thomas [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 12, 2006 11:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Input on Dundi On 12/12/06, Al Bochter [EMAIL PROTECTED] wrote: Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? We have been playing with DUNDi in a configuration similar to JR's whitepaper. Everything seems to be working fine but we have encountered a couple hurdles. Maybe others on the list have encountered these as well. 1.) When a registration server fails there doesn't seem to be an easy way to have clients automatically register to a new server. (our clients are mostly other asterisk boxes.) To solve this we are considering using DNS failover. Wow. I remember when I raised this as an issue I was accused of being a Asterisk heretic. The solution suggested was to, increased load not-withstanding, bring your phone registration period right down. Thanks for the tip Doug. Even when registration periods are set low, the clients don't know which new server to register to. It would be great to use SRV, however most of our clients are also Asterisk boxes, and as you know Asterisk does not support multiple SRV lookups. We have used DNS failover on other services in the past and have thought to try this with asterisk. Basicly our clients would register to FQDN like reg1.mydomain.com, then when that box fails we'd have DNS re-direct that name to a the IP of reg2.mydomain.com. This seems to work with web and ftp, but I'm not sure how asterisk will respond. Any thoughts? David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip help for newbie
www.asteriskguru.com On 12/12/06, blackwater dev [EMAIL PROTECTED] wrote: Does anyone know of any good step by step tutorials on getting sip set up? I have asterisk installed but can't seem to figure out how to get an account set up and connect from my xTen phone so I can try the demo. The tutorials I read online seem to go into voicepulse stuff and all and I don't have an account there so am a bit lost. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ASTCC and DTMF
I am having an issue with ASTCC that when callers call in and enter a number, asterisk will see more numbers than they entered (for instance if they enter 18005551212 asterisk may see 18005551221122). Anyone else see this ? Any work arounds ? Thanks. Dovid___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disregistering Constantly - message: chan_sip.c:11564 sip_poke_noanswer: Peer 'provider-13052181000' is now UNREACHABLE! Last qualify: 0
Hi guys, I configure one Fedora Core Linux 5 for use with asterisk as gateway using Digium TE110P interconected in Alcantel 4100 I've set up it to register 100 voip numbers on my provider. All calls on Alcatel is send to asterisk. In some periods of day i receive this messages on asterisk console: Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer 'provider-13052181000' is now UNREACHABLE! Last qualify: 0 Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer 'provider-13052181001' is now UNREACHABLE! Last qualify: 0 Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer 'provider-13052181002' is now UNREACHABLE! Last qualify: 0 In all 100 numbers. I already change the link, but the problem still happpen. I use in sip.conf have this configuration to register lines on provider: register=13052181000:[EMAIL PROTECTED]/13052181000 register=13052181001:[EMAIL PROTECTED]/13052181001 register=13052181002:[EMAIL PROTECTED]/13052181002 . . . [provider-13052181000] type=friend context=default secret=1221212 username=13052181000 host=sip.provider.com fromuser=13052181000 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes [provider-13052181001] type=friend context=default secret=1221212 username=13052181001 host=sip.provider.com fromuser=13052181001 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes [provider-13052181002] type=friend context=default secret=1221212 username=13052181002 host=sip.provider.com fromuser=13052181002 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes If i disable 30 lines and restartr asterisk all lines are register normaly. So, Have any limit in network stack or in asterisk ? Have any tunning that can i make on linux or in asterisk to resolve this question ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk manager
Hi i am trying to record a call with exten = 9,1,Answer exten = 9,2,Monitor exten = 9,3,Dial(SIP/200) This will record the call, but asterisk generates 2 files in /var/spool/asterisk/monitor/ -in.wav -out.wav Can i have only one file? Can i customize the path where to save the files? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conference between skinny user and many sip user
Hi, can i set up my asterisk for: - receive a skinny call in a specific context (yes, i have already compiled asteirsk with h323 support) - forward the call to a sip user A - make the sip user B join the call and create a conference between skinny caller, A and B maky thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [BULK] [asterisk-users] Asterisk manager
Try using MixMonitor instead. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nik600 Sent: Tuesday, December 12, 2006 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [BULK] [asterisk-users] Asterisk manager Importance: Low Hi i am trying to record a call with exten = 9,1,Answer exten = 9,2,Monitor exten = 9,3,Dial(SIP/200) This will record the call, but asterisk generates 2 files in /var/spool/asterisk/monitor/ -in.wav -out.wav Can i have only one file? Can i customize the path where to save the files? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk manager
Your line number nine should also specify a file name to monitor to and the
format, like this
exten = 9,2,Monitor(from-${CALLEDID}-at-${TIMESTAMP},wav)
or better yet, use MixMon instead, because this will merge the two files into
just one. (both sides of the call)
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nik600
Sent: Tuesday, December 12, 2006 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk manager
Hi
i am trying to record a call with
exten = 9,1,Answer
exten = 9,2,Monitor
exten = 9,3,Dial(SIP/200)
This will record the call, but asterisk generates 2 files in
/var/spool/asterisk/monitor/
-in.wav
-out.wav
Can i have only one file?
Can i customize the path where to save the files?
Thanks
___
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___
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RE: [asterisk-users] X100P clone dial problems.
Well I finally got it to work. I changed the DSL filter and it started to work. Very strange. Thanks for everyone's suggestions. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes Sent: Tuesday, 12 December 2006 5:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] X100P clone dial problems. Well, my PSTN card has: signalling-fxs_ks and that works for me. Klaverstyn, David C wrote: Thanks for your help. This is my file. [channels] language=au context=from-pstn signalling=fxo_ks ;rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no channel = 1 Upon reloading asterisk I get the following errors. Dec 11 19:03:45 WARNING[5265]: chan_zap.c:10874 setup_zap: Ignoring signalling Dec 11 19:03:45 ERROR[5265]: chan_zap.c:10305 setup_zap: Unable to reconfigure channel '1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes Sent: Monday, 11 December 2006 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] X100P clone dial problems. Klaverstyn, David C wrote: I have since added fxs_ks=1 This is meaningless. Follow the example that I posted. and channel = 1 This has not fixed the problem. I do notice a warning on the reload of asterisk. WARNING[4296]: chan_zap.c:10874 setup_zap: Ignoring signalling -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Monday, 11 December 2006 4:47 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] X100P clone dial problems. On Mon, Dec 11, 2006 at 04:18:20PM +1100, Klaverstyn, David C wrote: I'm not sure if I have a configuration problem or not. I am unable to dial out. When I try to dial in I can hear the phone ring on the dialling phone but Asterisk does not register anything. In zaptel.conf I have loadzone = au defaultzone=au fxsks=1 In zapata.conf language=au context=from-pstn Those need to be in the section [channels] and be followed by a channel = 1 to actually have any effect. You also must set signaling (signalling = fxs_ks; in your case). -- Howard. LANNet Computing Associates - Your Linux people http://lannetlinux.com When you want a computer system that works, just choose Linux; When you want a computer system that works, just, choose Microsoft. -- Flatter government, not fatter government; abolish the Australian states. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Howard. LANNet Computing Associates - Your Linux people http://lannetlinux.com When you want a computer system that works, just choose Linux; When you want a computer system that works, just, choose Microsoft. -- Flatter government, not fatter government; abolish the Australian states. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference between skinny user and many sip user
I think, that adhoc conferencing isn't possible in this way, instead you should use meetme, ie.: skinny user calls to user A and transfer his to meetme number skinny user calls to user B and transfer his to meetme number skinny user calls to meetme number all three speech in conference... nik600 wrote: Hi, can i set up my asterisk for: - receive a skinny call in a specific context (yes, i have already compiled asteirsk with h323 support) - forward the call to a sip user A - make the sip user B join the call and create a conference between skinny caller, A and B maky thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk manager
-= Info about application 'MixMonitor' =- [Synopsis] Record a call and mix the audio during the recording [Description] MixMonitor(file.ext[|options[|command]]) Records the audio on the current channel to the specified file. If the filename is an absolute path, uses that path, otherwise creates the file in the configured monitoring directory from asterisk.conf. nik600 wrote: Hi i am trying to record a call with exten = 9,1,Answer exten = 9,2,Monitor exten = 9,3,Dial(SIP/200) This will record the call, but asterisk generates 2 files in /var/spool/asterisk/monitor/ -in.wav -out.wav Can i have only one file? Can i customize the path where to save the files? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
I'm using 8.2.1 in 7961, it working fine, registration is OK, except I must disable qualify in asterisk (phone doesn't respond to qualify pings), one anoying bug removed is not displaying IP address of sip server (asterisk) in caller id, also same issue with needing rename jar*.sbn file on tftp server anybody made BLF working on 7961 (7970)? PJ Matt Gibson wrote: Hi All, Found out Cisco has some newer firmware available for the 7970 series of phones. New sip images are at version level 8.2 (instead of 8.0.2,8.0.3,8.0.4), posted Dec 10, 2006. This major jump in version numbers has fixed a few bugs (time zone not updating properly), but hasn't figured what some would consider to be showstoppers (registration not fully working, and mwi still not working). Just thought I would let you all know there's new firmware to mess around with! Also, to note, to get the phone to actually take this upgrade, and you're running your tftp server on a linux box, then you will need to rename one of the files for it to find it properly. # cd tftpdroot # mv jar70sip.8-2-0-55.sbn Jar70sip.8-2-0-55.sbn Calls in and Out work, though the phone still shows that dreaded red x next to the extension saying it's not registered. MWI is also still not working with 3 or 1 in the MWI indicator slot in the .xml file. And no, I won't email you the firmware, you need a cisco login to get one, so get a friend, or join cisco yourself! :) Happy Testing! Matt G http://www.voipphreak.ca ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Settings CallerId for outgoing calls based on the sip account making them
Hi, I have 10 DID numbers. Calls coming from the PSTN network are routed correctly to the SIP users based on the number that was called. But when sip users call the PSTN network, the CallerID should be set to correspondent with their DID number. At the moment I can set the CallerID to a global number, but I have no idea how to check who's making the call. All sip users start in the context [internal] Any ideas? Thank you. - 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. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
Hi Pavel, I tried to implicitly set qualify=no for the sip user, but am still seeing the registering icon for like 10 minutes on the screen of the 7970. It is actually registering, just the phone doesn't think it is. The phones always stay with a little red X on them showing the phone doesn't think it's registered. Weird. Thanks for the update! Hopefully these kick ass phones will work better soon! Matt G On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: I'm using 8.2.1 in 7961, it working fine, registration is OK, except I must disable qualify in asterisk (phone doesn't respond to qualify pings), one anoying bug removed is not displaying IP address of sip server (asterisk) in caller id, also same issue with needing rename jar*.sbn file on tftp server anybody made BLF working on 7961 (7970)? PJ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Recommendations for QoS, PoE Switches
Layer 2 switches support all the basic switching functionality. QoS, SNMP, POE, VLANs, Etc... depending on the model and features. Layer 3 switches are essentialy basic routers with a switch built in. One thing about Cisco CDP and a lot of POE switches is you can get CDP support with a custom Ethernet cable, just swap pins 4-5 with 7-8 (This is how I'm running Cisco 7940G's with a Dell POE Switch). - William J McCloskey Information Technology Manager [EMAIL PROTECTED] 503-827-8141 503-228-6747 fax www.timbercon.com - -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Monday, December 11, 2006 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Re: Recommendations for QoS, PoE Switches Edgewater Networks markets a 24 port switch, with PoE (both Cisco CDP and 802.3af supported), and Layer 2/3 management features that retails for less than $1500. The model is EC-2402POE-01 Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick May Sent: Monday, December 11, 2006 10:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Re: Recommendations for QoS, PoE Switches On Mon, Dec 11, 2006 at 09:53:53AM -0500, Zeeshan Zakaria wrote: What's the price for these HP switches? And also I someone can give me a link to some document where I can read about Layer 2 and Layer 3, how they help in VoIP traffic, it'll be helpful. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CDW's retail price was about $7,000. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SPA2100 sends an unexpected BYE message whentransmitting a FAX
Hi Mike, Do you have a full SIP trace? Cheers Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Aster Sent: Tuesday, 12 December 2006 11:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SPA2100 sends an unexpected BYE message whentransmitting a FAX Hi everyone, I'm trying to send a FAX with the following configuration: Analog FAX machine (OKI) -SPA21000-LAN-Asterisk PSTN I'm restricted to use passthru mode for faxing, instead of T.38 protocol, because the Asterisk box is running v1.2 and cannot be changed as it is in a heavy production environment. Anyway, it should work in passthru mode (G.711a) as the ATA and the Asterisk are in the same LAN with very low traffic. The problem arises when I try to send a fax: the Asterisk server initiates the call and, after a few seconds, the Linksys hangs the call by sending a BYE message: DEBUG[7416]: chan_sip.c:11375 handle_request: Received ACK (6) - Command in SIP ACK DEBUG[7416]: chan_sip.c:1396 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #258 DEBUG[7416]: chan_sip.c:1407 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 102: Match Found -- SIP read from 192.168.6.222:5060: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.6.222:5060;branch=z9hG4bK-6b0d80f5 From: 201 sip:[EMAIL PROTECTED];tag=c6bf15cfef0f4e11o0 To: sip:[EMAIL PROTECTED];tag=as715a2601 Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username=201,realm=asterisk,nonce=36741c67,uri=sip:[EMAIL PROTECTED] .6.220,algorithm=MD5,response=94f0139b69bb01ddc4aa362ab3edc130 User-Agent: Linksys/SPA2100-3.3.6 I'm using the following features: - Network jitter buffer: very high - Jitter buffer adjustment: disable - Call Waiting: no - 3 Way Calling: no - Echo Canceller: no - Silence suppression: no - Preferred Codec: G711a - Use pref. codec only: yes - Silence Threshold = medium - Echo Canc Enable = no - Echo Canc Adapt Enable = no - Echo Supp Enable = no - FAX CED Detect Enable = no - FAX CNG Detect Enable = no - FAX Passthru Codec = G711a - FAX Passthru Method = NSE - FAX Process NSE = yes - FAX Disable ECAN = no - FAX Codec Symmetric = yes - DTMF Tx Method = auto - Hook Flash Tx Method = none - Release Unused Codec = yes I have checked the SPA2100's logs, but I can't see anything of interest (and I couldn't find any documentation about this logs at Sipura's website). Has anyone suceed in sending a fax in a scenario like this? I would appreciate any help on this point. Best regards, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I am using the free version of SPAMfighter for private users. It has removed 3657 spam emails to date. Paying users do not have this message in their emails. Get the free SPAMfighter here: http://www.spamfighter.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Input on Dundi
1.) When a registration server fails there doesn't seem to be an easy
way to have clients automatically register to a new server. (our
clients are mostly other asterisk boxes.) To solve this we are
considering using DNS failover.
When registering with an Asterisk server to an Asterisk cluster of
servers, for the purpose of traversing a NAT or something else (to
solve a problem where direct contact cannot be performed), I would
suggest doing multiple registration to two registration servers, using
different names.
Like
registration [name1] to registration server 1
registration [name2] to registration server 2
in the outgoing dilaplan
exten = _NXXNXX,1,Dial(IAX2/server1..|j)
exten = _NXXNXX,102,Dial(IAX2/server2..
so if server one is not there the call will jump to the next server
or
exten = _NXXNXX,1,Dial(IAX2/server1IAX2/server2.
first server to answer will get the call.
you can do something similar calling from the cluster to the end Asterisk server
dundi lookup for [name1] if not available lookup [name2]
2.) If you plan to do any direct routing using the fullcontact
address like what is shown in JR's whitepaper, you may find that
fullcontact sometimes contains private network addresses. This makes
it impossible to route inbound calls directly to the client.
I recently started pulling the ipaddress and port from the database
instead of using the fullcontact field. Aaron Daniels helped me to
get the realtime query working instead of using the mysql connect
statements.
[lookupmysql]
include = invalid
exten = _X.,1,RealTime(sippeers|name|${EXTEN}|DN_)
exten = _X.,2,GotoIf($[${DN_ipaddr} = ]?${EXTEN},105:${EXTEN},3)
exten = _X.,3,Set([EMAIL PROTECTED]:${DN_port})
exten = _X.,4,Dial(SIP/${directdial},15,rj)
exten = _X.,5,Macro(sendtovm,${EXTEN})
exten = _X.,6,Hangup
exten = _X.,105,Macro(sendtovm,${EXTEN})
exten = _X.,106,Hangup
The RealTime command pulls all the entire record from the database and
prepends all the fields with the last argument (here is have DN_) so
when the record is pulled, all the records info is available as a
variable like DN_port and DN_ipaddr.
This is a really cool command. Hope this helps.
--
JR Richardson
Engineering for the Masses
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[asterisk-users] zapata.conf zaptel.conf
I am configuring two cards in Trixbox. 1 TE110P T-1 card and one TDM2400P with 16 fxs ports (All 24 show up in zaptel.conf so the PRI channels start at 25). Can I use a channel range to separate the config for each card, as shown below, or do I have to enter configs for each channel? Also in zaptel.conf I see that the TDM card is span 1 board zero, And the T-1 card is Span 2 board 0. So for the T-1 card I entered span=2,1,0,esf,b8zs Is this correct? What are the trailing 1,0 for after the span ID? OT-In reference to posting messages-what is top posting? -- ; Zapata telephony interface [trunkgroups] ; [channels] ; context=incoming switchtype=national ; signalling=fxs_ks usecallerid=yes cidsignalling=bell hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default channel=1-16 ; context=incoming switchtype=national ; signalling=pri_cpe usecallerid=yes cidsignalling=bell hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default channel=25-47 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Measuring VoIP latency and packet loss
Dear all, Are there anyone have ben to use some tool or method to measure latency and packet loss for VoIP packet ? - This email was sent using Student EEPIS-Webmail. http://student.eepis-its.edu/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] zapata.conf zaptel.conf
I am configuring two cards in Trixbox. 1 TE110P T-1 card and one TDM2400P with 16 fxs ports (All 24 show up in zaptel.conf so the PRI channels start at 25). Can I use a channel range to separate the config for each card, as shown below, or do I have to enter configs for each channel? Also in zaptel.conf I see that the TDM card is span 1 board zero, And the T-1 card is Span 2 board 0. So for the T-1 card I entered span=2,1,0,esf,b8zs Is this correct? What are the trailing 1,0 for after the span ID? Check out: http://www.voip-info.org/wiki/index.php?page=Zaptel.conf+span+syntax OT-In reference to posting messages-what is top posting? Top posting means putting your reply at the very top of your post and leaving the thread contents below. You'll notice that I left your original post mostly in tact, cutting out only the boring email header info. It is proper etiquette not to top post but instead put your replies at the very end of the post so that those reading it can see it in chronological order. If everyone top posted then the quoted thread would be in reverse chronological order and you'd need to scroll to the bottom to see the start of the discussion and then scroll up as you read. Most of us prefer to scroll down as we read! :) -- ; Zapata telephony interface [trunkgroups] ; [channels] ; context=incoming switchtype=national ; signalling=fxs_ks usecallerid=yes cidsignalling=bell hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default channel=1-16 ; context=incoming switchtype=national ; signalling=pri_cpe usecallerid=yes cidsignalling=bell hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default channel=25-47 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] caller ID authentication
Is there a utility or srcipt in asterisk which accepts calls based on caller ID and gives a busy signal if the caller ID is not on the list. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller ID authentication
Vernier Umali wrote: Is there a utility or srcipt in asterisk which accepts calls based on caller ID and gives a busy signal if the caller ID is not on the list. Thanks Search the Wiki or Mailing List archives for the ex-girlfriend option. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Input on Dundi
On 12/12/06, JR Richardson [EMAIL PROTECTED] wrote:
1.) When a registration server fails there doesn't seem to be an easy
way to have clients automatically register to a new server. (our
clients are mostly other asterisk boxes.) To solve this we are
considering using DNS failover.
When registering with an Asterisk server to an Asterisk cluster of
servers, for the purpose of traversing a NAT or something else (to
solve a problem where direct contact cannot be performed), I would
suggest doing multiple registration to two registration servers, using
different names.
Like
registration [name1] to registration server 1
registration [name2] to registration server 2
in the outgoing dilaplan
exten = _NXXNXX,1,Dial(IAX2/server1..|j)
exten = _NXXNXX,102,Dial(IAX2/server2..
so if server one is not there the call will jump to the next server
or
exten = _NXXNXX,1,Dial(IAX2/server1IAX2/server2.
first server to answer will get the call.
you can do something similar calling from the cluster to the end Asterisk server
dundi lookup for [name1] if not available lookup [name2]
2.) If you plan to do any direct routing using the fullcontact
address like what is shown in JR's whitepaper, you may find that
fullcontact sometimes contains private network addresses. This makes
it impossible to route inbound calls directly to the client.
I recently started pulling the ipaddress and port from the database
instead of using the fullcontact field. Aaron Daniels helped me to
get the realtime query working instead of using the mysql connect
statements.
[lookupmysql]
include = invalid
exten = _X.,1,RealTime(sippeers|name|${EXTEN}|DN_)
exten = _X.,2,GotoIf($[${DN_ipaddr} = ]?${EXTEN},105:${EXTEN},3)
exten = _X.,3,Set([EMAIL PROTECTED]:${DN_port})
exten = _X.,4,Dial(SIP/${directdial},15,rj)
exten = _X.,5,Macro(sendtovm,${EXTEN})
exten = _X.,6,Hangup
exten = _X.,105,Macro(sendtovm,${EXTEN})
exten = _X.,106,Hangup
The RealTime command pulls all the entire record from the database and
prepends all the fields with the last argument (here is have DN_) so
when the record is pulled, all the records info is available as a
variable like DN_port and DN_ipaddr.
This is a really cool command. Hope this helps.
Wow, thanks for the examples JR. This is exactly what I needed. I was
not aware of the RealTime command. That will be very useful.
David
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Re: [asterisk-users] Settings CallerId for outgoing calls based on thesip account making them
- Original Message - From: Timothy Parez [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, December 13, 2006 1:03 AM Subject: [asterisk-users] Settings CallerId for outgoing calls based on thesip account making them Hi, I have 10 DID numbers. Calls coming from the PSTN network are routed correctly to the SIP users based on the number that was called. But when sip users call the PSTN network, the CallerID should be set to correspondent with their DID number. At the moment I can set the CallerID to a global number, but I have no idea how to check who's making the call. All sip users start in the context [internal] Any ideas? Thank you. If your provider allows you to set your own CID you can try setting it in sip.conf. If not you can create a seperate context for each user and then have them use the diffrent DID's. Another option is too have diffrent pattern matches. For instance to use DID1 dial 1+number for DID2 dial 2+number etc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add include statement into Realtime static
Doug, How do you use it ? What do you insert in to the DB ? Thanks. Dovid - Original Message - From: Douglas Garstang To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Sent: Tuesday, December 12, 2006 5:42 PM Subject: RE: [asterisk-users] How to add include statement into Realtime static The 'include =' statement works fine for us in realtime static. Doug. -Original Message- From: Fran Oliveira [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 12, 2006 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [asterisk-users] How to add include statement into Realtime static you must use the switch command. I am not sure, but I think you should configure config realtime also, otherwise this command will be in extensions.conf Take a look in voip-info.org 2006/12/12, Tielin Xu [EMAIL PROTECTED]: Hi List: I can not find out an example how to store include = context name statement into Realtime static. Please help me on this one. Thanks, Tielin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help getting started with asterisk
I am new to asterisk. I need help getting started, if it's even worth getting started. I say if it's worth getting started because I'm not sure if my hardware will even work with asterisk. I have a US ROBOTICS 56K V.90 PCI SOFT MODEM. I have standard twisted pair telephone wire. I can't afford to alter my hardware. I know I won't be able to do any sophistocated VoIP stuff. All I want is for asterisk to provide caller ID information for my Gentoo box and to drop calls for certain phone numbers I specify. Someone on the Gentoo list told me that for the caller ID bit I should check my modem's manual to find out how; my modem did not come with a manual. I've emerged asterisk on my Gentoo system, as well as the zaptel driver package. I issued /etc/init.d/asterisk start and confirmed with ps that it was running. I got a command console with asterisk -r. I then called my home line that the computer is plugged into from my cell phone. I heard my wife's cordless ringing in the other room, and I heard some breaks on the phone line, but nothing else. I let it ring ten times. What would I have to do to get asterisk to realize that the PC is connected to the phone line? I know it is because I can dial out with kppp. Also, I've been trying to follow the AsteriskTFOT.pdf file. On page 79 it says to add a few lines to /etc/zaptel.conf and then modprobe wctdm. I did that. I then ran /sbin/ztcnf -vv to make sure everything was right. I got this: camille ~ # modprobe wctdm camille ~ # /sbin/ztcfg -vv Zaptel Configuration == Channel map: Channel 02: FXS Kewlstart (Default) (Slaves: 02) 1 channels configured. ZT_CHANCONFIG failed on channel 2: No such device or address (6) What does that mean? What device was it looking for? Please help! -Michael Sullivan- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip help for newbie
Thanks for the info, I've gone through the tutorial and followed it and asterisk is running but I just can't seem to log in. The xten phone just tells me connection timed out. I'm simply running asterisk on a webserver that is also running apache and service content. I simply pinged the box to get the ip to plug into the softphone. Do I need to open a port or something? On 12/12/06, Forrest Beck [EMAIL PROTECTED] wrote: www.asteriskguru.com On 12/12/06, blackwater dev [EMAIL PROTECTED] wrote: Does anyone know of any good step by step tutorials on getting sip set up? I have asterisk installed but can't seem to figure out how to get an account set up and connect from my xTen phone so I can try the demo. The tutorials I read online seem to go into voicepulse stuff and all and I don't have an account there so am a bit lost. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help getting started with asterisk
I would be very surprised if your modem is supported by Asterisk - but I suppose it's worth a try. What does 'zap show status' and 'zap show channels' show in the Asterisk CLI? PaulH On Tue, 2006-12-12 at 21:13 -0600, Michael Sullivan wrote: I am new to asterisk. I need help getting started, if it's even worth getting started. I say if it's worth getting started because I'm not sure if my hardware will even work with asterisk. I have a US ROBOTICS 56K V.90 PCI SOFT MODEM. I have standard twisted pair telephone wire. I can't afford to alter my hardware. I know I won't be able to do any sophistocated VoIP stuff. All I want is for asterisk to provide caller ID information for my Gentoo box and to drop calls for certain phone numbers I specify. Someone on the Gentoo list told me that for the caller ID bit I should check my modem's manual to find out how; my modem did not come with a manual. I've emerged asterisk on my Gentoo system, as well as the zaptel driver package. I issued /etc/init.d/asterisk start and confirmed with ps that it was running. I got a command console with asterisk -r. I then called my home line that the computer is plugged into from my cell phone. I heard my wife's cordless ringing in the other room, and I heard some breaks on the phone line, but nothing else. I let it ring ten times. What would I have to do to get asterisk to realize that the PC is connected to the phone line? I know it is because I can dial out with kppp. Also, I've been trying to follow the AsteriskTFOT.pdf file. On page 79 it says to add a few lines to /etc/zaptel.conf and then modprobe wctdm. I did that. I then ran /sbin/ztcnf -vv to make sure everything was right. I got this: camille ~ # modprobe wctdm camille ~ # /sbin/ztcfg -vv Zaptel Configuration == Channel map: Channel 02: FXS Kewlstart (Default) (Slaves: 02) 1 channels configured. ZT_CHANCONFIG failed on channel 2: No such device or address (6) What does that mean? What device was it looking for? Please help! -Michael Sullivan- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help getting started with asterisk
On Wed, 2006-12-13 at 14:21 +1100, Paul Hales wrote: I would be very surprised if your modem is supported by Asterisk - but I suppose it's worth a try. What does 'zap show status' and 'zap show channels' show in the Asterisk CLI? PaulH camille*CLI zap show status No Zaptel interface found. Dec 12 21:23:17 WARNING[26372]: chan_zap.c:9774 zap_show_status: Unable to open /dev/zap/ctl: Permission denied camille*CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault camille*CLI I checked the perms of /dev/zap/ctl: camille ~ # ls -l /dev/zap/ctl crw-rw 1 root dialout 196, 0 Dec 12 21:23 /dev/zap/ctl Is this not correct? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help getting started with asterisk
As Paul Hales said, I doubt that modem is supported. To interface with a regular phone line you'll need to get a supported card. You can read about it online. To just get started with playing, I recommend you go ahead with the sophistocated VoIP stuff.. Perhaps sign up with IPKALL or Stanaphone. Google will tell you all about how to connect them... Good luck and have fun! Todd On Dec 12, 2006, at 10:13 PM, Michael Sullivan wrote: I am new to asterisk. I need help getting started, if it's even worth getting started. I say if it's worth getting started because I'm not sure if my hardware will even work with asterisk. I have a US ROBOTICS 56K V.90 PCI SOFT MODEM. I have standard twisted pair telephone wire. I can't afford to alter my hardware. I know I won't be able to do any sophistocated VoIP stuff. All I want is for asterisk to provide caller ID information for my Gentoo box and to drop calls for certain phone numbers I specify. Someone on the Gentoo list told me that for the caller ID bit I should check my modem's manual to find out how; my modem did not come with a manual. I've emerged asterisk on my Gentoo system, as well as the zaptel driver package. I issued /etc/init.d/asterisk start and confirmed with ps that it was running. I got a command console with asterisk -r. I then called my home line that the computer is plugged into from my cell phone. I heard my wife's cordless ringing in the other room, and I heard some breaks on the phone line, but nothing else. I let it ring ten times. What would I have to do to get asterisk to realize that the PC is connected to the phone line? I know it is because I can dial out with kppp. Also, I've been trying to follow the AsteriskTFOT.pdf file. On page 79 it says to add a few lines to /etc/zaptel.conf and then modprobe wctdm. I did that. I then ran /sbin/ztcnf -vv to make sure everything was right. I got this: camille ~ # modprobe wctdm camille ~ # /sbin/ztcfg -vv Zaptel Configuration == Channel map: Channel 02: FXS Kewlstart (Default) (Slaves: 02) 1 channels configured. ZT_CHANCONFIG failed on channel 2: No such device or address (6) What does that mean? What device was it looking for? Please help! -Michael Sullivan- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller ID authentication
Thanks On 12/13/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Vernier Umali wrote: Is there a utility or srcipt in asterisk which accepts calls based on caller ID and gives a busy signal if the caller ID is not on the list. Thanks Search the Wiki or Mailing List archives for the ex-girlfriend option. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zapata.conf: cannot set txgain lower than -6.3 ?
Thanks for confirming that it can go that low. The decimal point doesn't make a difference; I just added a decimal to determine the threshold at which point things stop working (-7 doesn't work, but -6 does) Steve On 12/12/06, Andy Kuo [EMAIL PROTECTED] wrote: Hi Steve, I tried txgain as low as -18 without any problem, but I never tried anything with decimal points. Andy On 12/12/06, Steve Hsieh [EMAIL PROTECTED] wrote: Greetings everyone, I have a Digium TDM400P card with both an FXO and FXS module to connect to the phone company and to a standard phone. The problem is that the volume of my voice is going out too loud. I tried lowering the txgain value in zapata.conf to compensate, but all audio drops out completely if I set txgain to -6.4 or lower. If I set it to -6.3, then everything works (but still too loud). Is there a limitation as to how low txgain can be set? From voip-info.org, I was under the impression that the range was -100 to 100. Even the page at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf shows an example txgain of -15.9 Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help getting started with asterisk
What does zttool show? And after you 'modprobe wctdm' what does your dmesg read? /var/log/messages? You should see something about a card being recognised PaulH On Tue, 2006-12-12 at 21:24 -0600, Michael Sullivan wrote: On Wed, 2006-12-13 at 14:21 +1100, Paul Hales wrote: I would be very surprised if your modem is supported by Asterisk - but I suppose it's worth a try. What does 'zap show status' and 'zap show channels' show in the Asterisk CLI? PaulH camille*CLI zap show status No Zaptel interface found. Dec 12 21:23:17 WARNING[26372]: chan_zap.c:9774 zap_show_status: Unable to open /dev/zap/ctl: Permission denied camille*CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault camille*CLI I checked the perms of /dev/zap/ctl: camille ~ # ls -l /dev/zap/ctl crw-rw 1 root dialout 196, 0 Dec 12 21:23 /dev/zap/ctl Is this not correct? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: CLI History
And ofcourz, be careful, with your fingers on the CLI or elsewhere, esp on a production server. cheerz - Ben. Benny Amorsen wrote: DG == Douglas Garstang [EMAIL PROTECTED] writes: DG When I exited the CLI and re-entered and pressed ctrl-c, That's where your problem is. Use exit and not ctrl-c to leave asterisk -r. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help getting started with asterisk
I definitely agree with the reading thing - it's a great way to learn http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 PaulH On Tue, 2006-12-12 at 22:53 -0500, Todd- Asterisk wrote: As Paul Hales said, I doubt that modem is supported. To interface with a regular phone line you'll need to get a supported card. You can read about it online. To just get started with playing, I recommend you go ahead with the sophistocated VoIP stuff.. Perhaps sign up with IPKALL or Stanaphone. Google will tell you all about how to connect them... Good luck and have fun! Todd On Dec 12, 2006, at 10:13 PM, Michael Sullivan wrote: I am new to asterisk. I need help getting started, if it's even worth getting started. I say if it's worth getting started because I'm not sure if my hardware will even work with asterisk. I have a US ROBOTICS 56K V.90 PCI SOFT MODEM. I have standard twisted pair telephone wire. I can't afford to alter my hardware. I know I won't be able to do any sophistocated VoIP stuff. All I want is for asterisk to provide caller ID information for my Gentoo box and to drop calls for certain phone numbers I specify. Someone on the Gentoo list told me that for the caller ID bit I should check my modem's manual to find out how; my modem did not come with a manual. I've emerged asterisk on my Gentoo system, as well as the zaptel driver package. I issued /etc/init.d/asterisk start and confirmed with ps that it was running. I got a command console with asterisk -r. I then called my home line that the computer is plugged into from my cell phone. I heard my wife's cordless ringing in the other room, and I heard some breaks on the phone line, but nothing else. I let it ring ten times. What would I have to do to get asterisk to realize that the PC is connected to the phone line? I know it is because I can dial out with kppp. Also, I've been trying to follow the AsteriskTFOT.pdf file. On page 79 it says to add a few lines to /etc/zaptel.conf and then modprobe wctdm. I did that. I then ran /sbin/ztcnf -vv to make sure everything was right. I got this: camille ~ # modprobe wctdm camille ~ # /sbin/ztcfg -vv Zaptel Configuration == Channel map: Channel 02: FXS Kewlstart (Default) (Slaves: 02) 1 channels configured. ZT_CHANCONFIG failed on channel 2: No such device or address (6) What does that mean? What device was it looking for? Please help! -Michael Sullivan- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem in making outbound calls in PRI
Danny, You've got a lot of stuff in there that isn't used for a PRI/ISDN. Mine setup attached. This if for a T1, and in the US. Please make adjustments for your area: Thank you Doug ! This mess was an outcome of many forums and misunderstood concepts. Well, I have not tested with these, cause we are still running hardware PBX, which is being used in daytime. Will try this out ! Thanks for your advice. - Danny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem in making outbound calls in PRI
Hi Tim, #- If that doesn't help, please send us the output of cat /proc/zaptel/1 And enable pri debugging in asterisk and send the output when you make a call. Oh ! I did not know about this. *pri intense debug span* Don't be afraid to ask the provider what they are seeing... PRI outbound calls are working for sure. We are still using our hw pbx. I will test with your configs. First time I put in a PRI I spent 2 days messing with it before I rang them, to be told that they hadn't enabled outbound calls yet! T. Thank you, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller ID authentication
I looked at the ex-girlfriend option and it's just part of what I needed. What I do want is to setup a whitelist or numbers which can access the asterisk box and its extensions. All other numbers will be given a congestion or busy tone regardless of what extension they are trying to reach. It would be better that the whitelist is in an external database of list that asterisk can look up. On 12/13/06, Vernier Umali [EMAIL PROTECTED] wrote: Thanks On 12/13/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Vernier Umali wrote: Is there a utility or srcipt in asterisk which accepts calls based on caller ID and gives a busy signal if the caller ID is not on the list. Thanks Search the Wiki or Mailing List archives for the ex-girlfriend option. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
