Re: [Asterisk-Users] Dumb TDM400P question
Don't know if this helps, but my installed 4-port fxo card has the rj11 jack closest to the pci edge connector as zap/4, and the rj11 away from the pci edge connector as zap/1. The board is installed and working, so can't look at much more. I have a TDM400P with 3 fxs and 1 fxo ports. I need to know which phone connector corresponds to which module and also which port number. If we are looking at the card with the PCI connector at the bottom, fxs/o modules at the top and the phone jacks on the left - do the phone jacks start at the top or the bottom (top is port 1 or bottom is port 1)? And which phone jack belongs to which module (top phone = left module, or top phone = right module)? module a module b module c module d phone w phone x phone y phone z w=a, x=b, y=c, z=d or w=d, x=c, y=b, z=a or something else? and then is w=1, x=2, y=3, z=4 or z=1, y=2, x=3, w=4 or something else? TIA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP error dialing
Phillip group, I tried what you suggested and it did not work i included some more information for you to take a look at... i have got the MGCP working sort of for my asterisk server. My phone plugged into the dlink gateway does not ring when i call it. My sip phone does ring when i dial the extention. asterisk CLI shows its ringing correctly. I am using the dlink gateway whch has 2 ports in it. I have an extention for both ports in my extentions.conf. Asterisk appears to reconize that a phone is in there as it does not go to voice mail right away and it shows its dialing in the command line interface. I tried calling aaln/1 when no phone was in there and it went right to voice mail. I plug a phone in and it just says ringing. Below is my conf that i have now.Is there anything I need to configure in the Dlink gateway for this to work with asterisk? my gateway works fine and i use it normally for calls. I might have missed something very simple but I never tried this before so i am not sure... Dlink gateway Wan port to switch, Lan port no cable in it. line 1, normal phone plugged in *CLI show version Asterisk 0.9.0 built by [EMAIL PROTECTED] on a i686 running Linux *CLI extentions.conf [default] exten = 2002,1,Dial(MGCP/aaln/[EMAIL PROTECTED]) exten = 2002,2,Hangup mgcp.conf [general] port=2427 ;bindaddr= [10.0.1.150] host=10.0.1.150 canreinvite=no context=default line = aaln/1 asterisk message output when you call the phone -- Executing Dial(SIP/2204-ac95, MGCP/aaln/[EMAIL PROTECTED]) in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: 0, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Called aaln/[EMAIL PROTECTED] -- MGCP/aaln/[EMAIL PROTECTED] is ringing asterisk output when you run asterisk -vvvgc chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf': Found -- Allocating subchannel '0' on aaln/[EMAIL PROTECTED] -- Allocating subchannel '1' on aaln/[EMAIL PROTECTED] -- Added gateway '10.0.1.150' == MGCP Listening on 0.0.0.0:2427 == Using TOS bits 0 == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) Warning, flexibel rate not heavily tested! MGCP Auditing endpoint aaln/[EMAIL PROTECTED] for hookstate when I run mgcp show endpoints *CLI mgcp show endpoints Gateway '10.0.1.150' at 10.0.1.150 (Static) -- 'aaln/[EMAIL PROTECTED] in 'default' is idle when I run mgcp audit endpoint aaln/[EMAIL PROTECTED] CLI mgcp audit endpoint aaln/[EMAIL PROTECTED] Posting Request: AUEP 6 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A,R,D,S,X,N,I,T,O,ES,VS,E,MD,M to 10.0.1.150:2427 May 22 03:40:13 NOTICE[1074449120]: chan_mgcp.c:417 mgcp_postrequest: Timeout waiting for response to message:1, lastouttime: 1085211476, now: 1085211613. Dumping pending queue May 22 03:40:13 NOTICE[1074449120]: chan_mgcp.c:398 dump_queue: Removing message from aaln/[EMAIL PROTECTED] tansaction 1 May 22 03:40:13 NOTICE[1074449120]: chan_mgcp.c:398 dump_queue: Removing message from aaln/[EMAIL PROTECTED] tansaction 2 May 22 03:40:13 NOTICE[1074449120]: chan_mgcp.c:398 dump_queue: Removing message from aaln/[EMAIL PROTECTED] tansaction 3 May 22 03:40:13 NOTICE[1074449120]: chan_mgcp.c:398 dump_queue: Removing message from aaln/[EMAIL PROTECTED] tansaction 4 May 22 03:40:13 NOTICE[1074449120]: chan_mgcp.c:398 dump_queue: Removing message from aaln/[EMAIL PROTECTED] tansaction 5 Any ideas? I followed wiki and any docs I can find on mgcp with the box.Everything else works... do i need to have something in bindaddr= for mgcp.conf? I marked that out. steven kalcevich Quoting Philipp von Klitzing [EMAIL PROTECTED]: Hi! I am trying to dial a mgcp extention from my sip phone and i am getting this error message. anyone got any idea? Do a mgcp show endpoints at the CLI and watch the output. May 19 22:30:01 NOTICE[1251156800]: chan_mgcp.c:1104 find_subchannel: Gateway '10.0.1.150' (and thus its endpoint 'aaln/1') does not exist May 19 22:30:01 WARNING[1251156800]: chan_mgcp.c:2608 mgcp_request: Unable to find MGCP endpoint 'aaln/[EMAIL PROTECTED]' mgcp.conf [dlinkgw] host=10.0.1.150 canreinvite=no context=default line = aaln/1 Change [dlinkgw] to [10.0.1.150], and the do a restart - depending on the Asterisk CVS version that you are using a reload or mgcp reload might not be sufficent/ might not work. See also: http://www.voip-info.org/wiki-Asterisk+config+mgcp.conf Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Steve Kalcevich Commercial Accounts Primus Telecommunications Canada Inc. Direct: 416-207-4613 Toll Free: 1-888-502-8380, ext. 8313 Fax: 1-800-861-3035 E
RE: [Asterisk-Users] Fedora Core 2 and Kernel 2.6
Really you should link /usr/src/linux-2.6 to /lib/modules/`uname -r`/build then you don't have to do anything special and it'll build... That directory and all the files in it are installed by the kernel rpm, you don't even need kernel-source for it... Although I haven't tried compiling without it installed I patched my zaptel Makefile to just reference that directory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua M. Thompson Sent: Thursday, May 20, 2004 5:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fedora Core 2 and Kernel 2.6 On Thu, 2004-05-20 at 05:12, WipeOut wrote: When trying to build zaptel it required me to link /usr/scr/linux-2.6 to the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :) After than I tried again but the page rolls with errors and finally ends with.. make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-1.358' make: *** [linux26] Error 2 Anyone got ant ideas? You'll need to configure the source tree before zaptel will compile. The config files are in /usr/src/linux-2.6/configs...copy the one that matches what you're running to /usr/src/linux-2.6/.config and then run make oldconfig. Zaptel should compile after that. -- Joshua M. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
Re: [Asterisk-Users] Asterisk and OH323
this options remove first number
try
exten = _0.,1,Dial(h323/${EXTEN:[EMAIL PROTECTED]:1)
Hekuran Doli wrote:
Need to anounce that Im using sip to h323!
Is there any beter solution to do this ?
.
Can you tell us in details what the problem is (or I didnt understand)?
if the problem is on call forwardin you have to add the following line
on the context you are using:
exten = _0.,1,Dial(h323/${EXTEN:[EMAIL PROTECTED]) so all calls
starting with with 0 will be forwardert to the gatekeeper`s IP adress
(gatekeepers-IP-address) Note:I use this for international calls so If I
want to dial 37744387555 I use 037744387555.
with worked for me.
H323: You dont have to add something special to h323.conf you can find a
sample of h323.conf on /usr/src/asterisk/channels/h323 and just need to
enable: gatekeeper = gatekeepers-IP-adress
Hope you`ll find this useful!
Best Regards
Hekuran Doli
Hello,
i want to use asterisk as a gateway for H323-Phones.
But i cant get it work.
I'm using a gatekeeper on another computer. My IP-phone is registered
there.
Does anybody can sent me an oh323.conf and extension.conf as examples?
Thanks in advance
Erik Bastian
--
NEU : GMX Internet.FreeDSL
Ab sofort DSL-Tarif ohne Grundgebühr: http://www.gmx.net/dsl
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[Asterisk-Users] Re: G.729a beta codec on old Pentiums
Hi, new codec runs with snom 200 ! greetings nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan CAPI and Latest CVS wont compile
When I saw the update for Cisco Phone RTP issue I thought I would try it. Unfortunately chan_capi wont compile on this update. Can anyone recommend a good * release for Capi, Bri ISDN and Cisco 7940s SIP 6.3. Or will CHAN_CAPI also be updated ? Running Eicon Diva Bri Cards. Error: chan_capi.c:1187: too many arguments to function ast_dsp_process
Re: [Asterisk-Users] Chan CAPI and Latest CVS wont compile
http://lists.digium.com/pipermail/asterisk-users/2004-April/044125.html - Original Message - From: Craig Waddington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 22, 2004 12:24 PM Subject: [Asterisk-Users] Chan CAPI and Latest CVS wont compile When I saw the update for Cisco Phone RTP issue I thought I would try it. Unfortunately chan_capi wont compile on this update. Can anyone recommend a good * release for Capi, Bri ISDN and Cisco 7940's SIP 6.3. Or will CHAN_CAPI also be updated ? Running Eicon Diva Bri Cards. Error: chan_capi.c:1187: too many arguments to function 'ast_dsp_process' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoicePulse SIP
H - can anybody confirm this. I have generally had little luck with IAX in any case so I must admit I assumed (due to info from www.voip-info.org) that it was due to lack of timing device. I have actually not tried to do any trunking - just normal calls. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris A. Icide Sent: 22 May 2004 13:26 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoicePulse SIP Lars, I could be quite wrong, but I think you only need a 'timing' source if you want to use trunking over IAX. You can still use IAX without trunking if you don't have any sort of timing device. -Chris On 06:39 PM 5/21/2004, Lars Boegild Thomsen wrote: Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Chan CAPI and Latest CVS wont compile
Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chrétien Wetemans Sent: 22 May 2004 12:19 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Chan CAPI and Latest CVS wont compile http://lists.digium.com/pipermail/asterisk-users/2004-April/044125.html - Original Message - From: Craig Waddington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 22, 2004 12:24 PM Subject: [Asterisk-Users] Chan CAPI and Latest CVS wont compile When I saw the update for Cisco Phone RTP issue I thought I would try it. Unfortunately chan_capi wont compile on this update. Can anyone recommend a good * release for Capi, Bri ISDN and Cisco 7940's SIP 6.3. Or will CHAN_CAPI also be updated ? Running Eicon Diva Bri Cards. Error: chan_capi.c:1187: too many arguments to function 'ast_dsp_process' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dial application - continue in context
Hi! I'm tring to do some DB operations before and after a call. I see the 'g' option in dial to continue in context if the destination hangs up, but what if the originator hangs up? You either need to run a CRON job for this clean up, or do that at the beginning of the next call - whatever suits you better. Note: The h extension is not reliable enough to solve your problem. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP error dialing
Hi! Below is my conf that i have now.Is there anything I need to configure in the Dlink gateway for this to work with asterisk? Here a few things you can try: - upgrade to CVS-HEAD (not 0.9.0) and see if things are different - issue a ngrep port 2727 to monitor what your dlink is sending - uncomment the bindaddr= statement Make sure you do a RESTART and not a RELOAD after any changes that are supposed to affect MGCP. If you continue to experience problems please open a bug report and include as much data as you can provide. In this case you might also want to try to go back to CVS HEAD of 03/05/04 00:50:56. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call waiting indicator do not work for me.
Hi, The call waiting indicator do not work for me. I am using a snom200 cwi is switched on in phone-config. Have asked snom, but there are can not help me, because it is working for them. When it is coming in an call while the phone is busy. The phone returns: -- Got SIP response 486 Busy Here back from 190.100.200.19 But it should not, should make a call waiting indication. (The same behaviour is when i am dialing the phone (in idle) from extern without making an exten = s,x,Answer.) greeting nicolas SIP/2.0 100 Trying Via: SIP/2.0/UDP 190.100.200.18:5060;branch=z9hG4bK-kcw3axoq8oei From: Astrid Buero sip:[EMAIL PROTECTED];tag=g8uj4z79n7 To: sip:[EMAIL PROTECTED];user=phone;intercom=true;tag=as30cdf7be Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 190.100.200.18:5060 -- Executing Dial(SIP/200-409e, SIP/101|60|Ttr) in new stack We're at 190.100.200.1 port 16492 Answering with preferred capability 1024 Answering with preferred capability 8 Answering with preferred capability 256 Answering with preferred capability 2 Answering with preferred capability 1 Answering with preferred capability 4 Answering with preferred capability 128 Answering with non-codec capability 1 12 headers, 16 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK5a5bb490 From: Astrid Buero sip:[EMAIL PROTECTED];tag=as73047910 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sat, 22 May 2004 10:08:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 364 v=0 o=root 32409 32409 IN IP4 190.100.200.1 s=session c=IN IP4 190.100.200.1 t=0 0 m=audio 16492 RTP/AVP 97 8 18 3 4 0 7 101 a=rtpmap:97 iLBC/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:7 LPC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 190.100.200.19:5060 -- Called 101 Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 190.100.200.18:5060;branch=z9hG4bK-kcw3axoq8oei From: Astrid Buero sip:[EMAIL PROTECTED];tag=g8uj4z79n7 To: sip:[EMAIL PROTECTED];user=phone;intercom=true;tag=as30cdf7be Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 190.100.200.18:5060 alberspilnx8*CLI Sip read: SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK5a5bb490 From: Astrid Buero sip:[EMAIL PROTECTED];tag=as73047910 To: sip:[EMAIL PROTECTED];tag=7jlddlf13r Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:5060;line=lhynyb3y Content-Length: 0 8 headers, 0 lines -- Got SIP response 486 Busy Here back from 190.100.200.19 Transmitting:CLI ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK5a5bb490 From: Astrid Buero sip:[EMAIL PROTECTED];tag=as73047910 To: sip:[EMAIL PROTECTED];tag=7jlddlf13r Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 190.100.200.19:5060 -- SIP/101-8b54 is busy == Everyone is busy at this time -- Executing Wait(SIP/200-409e, 2) in new stack -- Executing VoiceMail(SIP/200-409e, u200) in new stack We're at 190.100.200.1 port 18090 Answering with preferred capability 1024 Answering with preferred capability 8 Answering with preferred capability 256 Answering with preferred capability 2 Answering with preferred capability 1 Answering with preferred capability 4 Answering with preferred capability 128 Answering with non-codec capability 1 Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 190.100.200.18:5060;branch=z9hG4bK-kcw3axoq8oei From: Astrid Buero sip:[EMAIL PROTECTED];tag=g8uj4z79n7 To: sip:[EMAIL PROTECTED];user=phone;intercom=true;tag=as30cdf7be Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 364 v=0 o=root 32409 32409 IN IP4 190.100.200.1 s=session c=IN IP4 190.100.200.1 t=0 0 m=audio 18090 RTP/AVP 97 8 18 3 4 0 7 101 a=rtpmap:97 iLBC/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:7 LPC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse SIP
I'm using Coloco now, which so far is working well. Where companies like VoicePulse buy services from a patchwork of CLECs in order to cover their markets, Coloco is a CLEC. The upside is that you cut out the middleman. But if you need a number in an area they don't serve you'll need to find a different provider. Coloco serves latas 236 and 238 (NPAs 301,240,410,443,703), which works well for me since I'm in 238. If you need numbers local to DC and central Maryland give them a shout (coloco.com). I hear they're also working with some other CLECs to get numbers in other areas but I don't have any details on that. -brian David H Hickman wrote: Who do you use now? David Hickman TSG Computer Consulting - Auctions 314-865-4752 x2 On May 21, 2004, at 8:49 PM, Brian Cuthie wrote: SIP used to work fine with VoicePulse. But the funny thing is I could never detect any signs that they were doing call accounting. I could make IAX calls and see them show up in the CDR and the $$ deducted from my account balance. But when I made SIP calls they appeared, by all measures, to be free. I wrote to their support department several times about this and never received a response. But that was pretty much par for the course with those guys so I moved on to another provider. -brian Lars Boegild Thomsen wrote: Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dial application - continue in context
Hi Phillip, It needs to occur right after the call. I'm tring to apply a sort of fromdomain call limit. So I need to keep track of how many are currently active -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Saturday, May 22, 2004 6:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dial application - continue in context Hi! I'm tring to do some DB operations before and after a call. I see the 'g' option in dial to continue in context if the destination hangs up, but what if the originator hangs up? You either need to run a CRON job for this clean up, or do that at the beginning of the next call - whatever suits you better. Note: The h extension is not reliable enough to solve your problem. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rejected NOTIFY requests
When I enable NOTIFY messages in my SIP device (Sipura), Asterisk reports: handle_request: Unknown SIP command 'NOTIFY' from 'xxx.xxx.xxx.xxx' When I disable NOTIFY messages, * reports the device UNREACHABLE, followed by REACHABLE every couple of minutes. I think I want NOTIFY on, because the Sipura is behind a NAT server, but the constant stream of warnings from * make me think I'm doing something wrong. Anyone have any ideas? Thanks in advance! Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rejected NOTIFY requests
At 7:18 AM -0700 on 5/22/04, Bruce Komito wrote: When I enable NOTIFY messages in my SIP device (Sipura), Asterisk reports: handle_request: Unknown SIP command 'NOTIFY' from 'xxx.xxx.xxx.xxx' When I disable NOTIFY messages, * reports the device UNREACHABLE, followed by REACHABLE every couple of minutes. I think I want NOTIFY on, because the Sipura is behind a NAT server, but the constant stream of warnings from * make me think I'm doing something wrong. Anyone have any ideas? Thanks in advance! Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 Try turning NOTIFY off, and adding qualify=3000 to your SIP stanzas for that host. This will cause Asterisk to originate a SIP OPTIONS query to the device every 60 seconds, and if the response takes more than 3000ms (3s) to return, then it will list it as unreachable. Otherwise, it will stay listed as 'reachable' and the NAT mappings will stay in place for the Sipura device since there will be traffic flowing at a reasonable rate between the server and the Sipura. It's probably the case that the NAT mapping for the firewall/NAT you're behind is less than the interval at which the Sipura sends NOTIFY requests, though I'm interested as to why it's reported as unreachable instead of unknown. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoicePulse SIP
Brian Cuthie wrote: I'm using Coloco now, which so far is working well. Where companies like VoicePulse buy services from a patchwork of CLECs in order to cover their markets, Coloco is a CLEC. The upside is that you cut out the middleman. But if you need a number in an area they don't serve you'll need to find a different provider. Coloco serves latas 236 and 238 (NPAs 301,240,410,443,703), which works well for me since I'm in 238. If you need numbers local to DC and central Maryland give them a shout (coloco.com). I hear they're also working with some other CLECs to get numbers in other areas but I don't have any details on that. -brian Is all above AFTER or BEFORE coloco is sent many emails asking please I would like to buy from your company? My experience with them is EXACTLY that!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse SIP
Welcome to Voicepulse and their lack of jitter buffer. This is the cause of your horrible sound. Will be just as bad with SIP. Which providers give you a jitter buffer? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Failed to write frame when pressing 'o'
I recently upgraded to the latest CVS and when a caller presses 'o' in voicemail, I get listed below. I have searched the archive for a suggestion and pared the sip.conf and extensions.conf to bare minimum to duplicate this scenario. Any suggestions? == Parsing '/etc/asterisk/enum.conf': == Parsing '/etc/asterisk/enum.conf': Found -- Registered to '198.22.67.70', who sees us as 67.86.244.235:4569 -- Executing VoiceMail2(SIP/2204-b2af, u2203) in new stack -- Playing 'voicemail/default/2203/greet' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Executing Dial(SIP/2204-b2af, SIP/2299) in new stack -- Called 2299 May 22 11:16:11 WARNING[1217669936]: chan_sip.c:1593 sip_write: Asked to transmit frame type 2, while native formats is 4 (read/write = 4/4) May 22 11:16:11 WARNING[1217669936]: file.c:539 ast_readaudio_callback: Failed to write frame == Spawn extension (local, o, 1) exited non-zero on 'SIP/2204-b2af' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to share Zap channels in 2 Asterisk servers
Hello I am trying to setup Asterisk on 2 servers PBX300 and PBX200. PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap device. Softphone from PBX200 can talk to softphone on PBX300 but no outgoing call from PBX200. I can call from PBX300 outside but I am unable to configure soft Phone defined in PBX200 to dial out side using PBX300 Zap devices. I am geting error message Rejected connect attempt from PBX200. Please help if this is possible. Thanks Deepak This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk cpu load
Hi ! Running asterisk (cvs 20/05/04) with config intel 2.4GHz(no SMP); 1GB RAM; 1*E100P; IAX1 slinear; RH 9-2.4.20-8(no patches) When running load (30 simultaneous calls), the server utilizes approx 10% CPU, but every 20-30 seconds it's a short peek where the asterisk-process takes 99% CPU. Have searched the mailing-list but not found any explanation/solution to the CPU-load peek... Can anybody advice me here ? Br / Jan Terje Tnnessen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamic SIP.CONF
Darren Nay wrote:
We are looking to expand our usage of Asterisk and I am trying to make as
much of the configuration dynamic as I possibly can. The only part that I'm
having problems with is sip.conf. I can get asterisk to register each
extension with our local SER SIP proxy dynamically by using the
sipfriends table in the database, but I'm having trouble with the message
waiting indicators (ie. SIP NOTIFY packets when a new voicemail is waiting).
-SNIP-
Is there a way to make this dynamic so that I don't have to add this into
sip.conf -every- single time that I add a new extension?
Only by extending the functionality of sip friends to include this extra
field...
I wouldn't bother doing this as ast_data (formally res_data) is being
developed to replace sip/iax friends.
If you want to take a sneak preview at this then see:
http://svn.asteriskdocs.org/res_data/ast_data/
I tried the following, but it didn't work ..
[default]
type=peer
host=dynamic
dtmfmode=inband
username=${EXTEN}
Mailbox=${EXTEN}
Am I on the right track, or way off base? :-)
Way off base ;)
That kind of syntax only works in extensions.conf
F
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[Asterisk-Users] fwd on busy when calling multiple extensions at once
Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3 phones ring on the first extension, but the dispatcher who is on a call, her phone does not ring. I want her second extension ring along with the other 3 phones first extensions. In sip.conf I have all the extensions set to incominglimit=1 and the pertinent part of extensions.conf is: exten = s,1,Dial(SIP/5000SIP5001SIP5002SIP5003,20,tr) exten = s,2,Dial(SIP/5004SIP5005SIP5006SIP5007,20,tr) and so on. If anybody has any insight, or a better solution, that would be great. Thanks, -Tor Roberts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFAX generates no tiff file
Hi, I am trying to receive a fax with the spandsp library. The sending fax says success but there is no tiff file generated. I use exten = 7000,1,rxfax(/tmp/testfax.tif) in my extensions.conf. The connection is via SIP/G.711 as I have read on the list that this can sometimes work (I know Fax over IP is troublesome without T.38). I think the transmission should not be the problem because of the success on the sending fax. This is the debug output. Am I missing something? TIA, Mike *CLI-- Executing RxFAX(SIP/uid-c5b6, /tmp/testfax.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up Slow carrier down T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamic SIP.CONF
So I've been kind of struggling with the notion of making my Asterisk
implementations dynamic, too. While I'd like to make everything directly
database driven, I'm not sure Asterisk is quite there yet.
I've been thinking of writing something that creates appropriate
configuration files from the database on a periodic basis, and then does
an Asterisk reload. This would introduce a small delay into
configuration changes, but it does have other benefits such as
decoupling the design of the database from Asterisk.
Any thoughts?
-brian
Fran Boon wrote:
Darren Nay wrote:
We are looking to expand our usage of Asterisk and I am trying to
make as
much of the configuration dynamic as I possibly can. The only part
that I'm
having problems with is sip.conf. I can get asterisk to register each
extension with our local SER SIP proxy dynamically by using the
sipfriends table in the database, but I'm having trouble with the
message
waiting indicators (ie. SIP NOTIFY packets when a new voicemail is
waiting).
-SNIP-
Is there a way to make this dynamic so that I don't have to add this
into
sip.conf -every- single time that I add a new extension?
Only by extending the functionality of sip friends to include this
extra field...
I wouldn't bother doing this as ast_data (formally res_data) is being
developed to replace sip/iax friends.
If you want to take a sneak preview at this then see:
http://svn.asteriskdocs.org/res_data/ast_data/
I tried the following, but it didn't work ..
[default]
type=peer
host=dynamic
dtmfmode=inband
username=${EXTEN}
Mailbox=${EXTEN}
Am I on the right track, or way off base? :-)
Way off base ;)
That kind of syntax only works in extensions.conf
F
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Re: [Asterisk-Users] VoicePulse SIP
[EMAIL PROTECTED] wrote: Welcome to Voicepulse and their lack of jitter buffer. This is the cause of your horrible sound. Will be just as bad with SIP. Which providers give you a jitter buffer? In Europe: VoipTalk and Magrathea. In the US: Iconnecthere. I am sure there are more. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse SIP
Lars Boegild Thomsen wrote: H - can anybody confirm this. I have generally had little luck with IAX in any case so I must admit I assumed (due to info from www.voip-info.org) that it was due to lack of timing device. I have actually not tried to do any trunking - just normal calls. That is correct. You only need it for IAX2 trunking. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fwd on busy when calling multiple extensions at once
You might consider using the Cisco SIP phones. They're smart enough to accept incoming calls for as many call appearances you have with the same SIP registration. -brian Tor Roberts wrote: Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3 phones ring on the first extension, but the dispatcher who is on a call, her phone does not ring. I want her second extension ring along with the other 3 phones first extensions. In sip.conf I have all the extensions set to incominglimit=1 and the pertinent part of extensions.conf is: exten = s,1,Dial(SIP/5000SIP5001SIP5002SIP5003,20,tr) exten = s,2,Dial(SIP/5004SIP5005SIP5006SIP5007,20,tr) and so on. If anybody has any insight, or a better solution, that would be great. Thanks, -Tor Roberts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to share Zap channels in 2 Asterisk servers
Call the PBX300 using IAX2 from PBX200, make sure that the call goes into the context that allows dial out. Example. exten = _543219XX,1,StripMSD,5 exten = _9XX,2,Dial/[EMAIL PROTECTED]/BYEXTENSION The first line looks for an access code '54321' followed by the access code for an outside line '9' and then a number. You next strip the access code for IAX linking and pass the rest to the other Asterisk PBX. The Asterisk PBX then runs the exten as if on the local machine. Simple huh. There is most likely a simpler method, but this works for me. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 22 May 2004 16:40 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] How to share Zap channels in 2 Asterisk servers Hello I am trying to setup Asterisk on 2 servers PBX300 and PBX200. PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap device. Softphone from PBX200 can talk to softphone on PBX300 but no outgoing call from PBX200. I can call from PBX300 outside but I am unable to configure soft Phone defined in PBX200 to dial out side using PBX300 Zap devices. I am geting error message Rejected connect attempt from PBX200. Please help if this is possible. Thanks Deepak This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rejected NOTIFY requests
John Todd wrote: At 7:18 AM -0700 on 5/22/04, Bruce Komito wrote: When I enable NOTIFY messages in my SIP device (Sipura), Asterisk reports: handle_request: Unknown SIP command 'NOTIFY' from 'xxx.xxx.xxx.xxx' When I disable NOTIFY messages, * reports the device UNREACHABLE, followed by REACHABLE every couple of minutes. I think I want NOTIFY on, because the Sipura is behind a NAT server, but the constant stream of warnings from * make me think I'm doing something wrong. Anyone have any ideas? Thanks in advance! Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 Try turning NOTIFY off, and adding qualify=3000 to your SIP stanzas for that host. This will cause Asterisk to originate a SIP OPTIONS query to the device every 60 seconds, and if the response takes more than 3000ms (3s) to return, then it will list it as unreachable. Otherwise, it will stay listed as 'reachable' and the NAT mappings will stay in place for the Sipura device since there will be traffic flowing at a reasonable rate between the server and the Sipura. It's probably the case that the NAT mapping for the firewall/NAT you're behind is less than the interval at which the Sipura sends NOTIFY requests, though I'm interested as to why it's reported as unreachable instead of unknown. Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk can handled the NAT traversal all by itself with Qualify (as John points out) disabling the NOTIFY will not change anything. The NOTIFY will in no way affect the status - unreachable/reachable. Another problem with the SIPURA is the lack of a working STUN solution. Even Grandstream works better with NAT today. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] rejected NOTIFY requests
Another problem with the SIPURA is the lack of a working STUN solution. Even Grandstream works better with NAT today. /O I second that!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamic SIP.CONF
What I do is have a cron with a perl script that recreates a sip-db.conf
(which has an #include in the sip.conf) then do a sip reload.
I's rather simple, all you have to write a temp and diff, if its
changed replace and reload, if not don't do a thing.
Do the same with extensions
On Sat, 2004-05-22 at 12:39, Fran Boon wrote:
Darren Nay wrote:
We are looking to expand our usage of Asterisk and I am trying to make as
much of the configuration dynamic as I possibly can. The only part that I'm
having problems with is sip.conf. I can get asterisk to register each
extension with our local SER SIP proxy dynamically by using the
sipfriends table in the database, but I'm having trouble with the message
waiting indicators (ie. SIP NOTIFY packets when a new voicemail is waiting).
-SNIP-
Is there a way to make this dynamic so that I don't have to add this into
sip.conf -every- single time that I add a new extension?
Only by extending the functionality of sip friends to include this extra
field...
I wouldn't bother doing this as ast_data (formally res_data) is being
developed to replace sip/iax friends.
If you want to take a sneak preview at this then see:
http://svn.asteriskdocs.org/res_data/ast_data/
I tried the following, but it didn't work ..
[default]
type=peer
host=dynamic
dtmfmode=inband
username=${EXTEN}
Mailbox=${EXTEN}
Am I on the right track, or way off base? :-)
Way off base ;)
That kind of syntax only works in extensions.conf
F
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--
Pablo Endres [EMAIL PROTECTED]
ComVoz Comunications
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[Asterisk-Users] loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop
Afternoon all, I'm trying to load Asterisk, however I am getting the following error: [skipping res_musiconhold.so] [chan_sip.so]May 22 18:42:24 WARNING[16384]: loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop May 22 18:42:24 WARNING[16384]: loader.c:421 load_modules: Loading module chan_sip.so failed! I've tried doing res_musiconhold.so=no in modules.conf with no change. I'm using a box without ztdummy or hardware, so I have no timing sources. I have configured sip.conf as well. I've done a search on google and the mailing list, and the only reference to this I could find was this post: http://lists.digium.com/pipermail/asterisk-users/2004-April/044507.html Which didn't really give me a whole lot more to go on than what I already know... I'm sure this is a simple problem which I'll smack my forehead when I hear, but so far it escapes me. Thanks, Leif Madsen. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop
Leif Madsen wrote: Afternoon all, I'm trying to load Asterisk, however I am getting the following error: [skipping res_musiconhold.so] [chan_sip.so]May 22 18:42:24 WARNING[16384]: loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop May 22 18:42:24 WARNING[16384]: loader.c:421 load_modules: Loading module chan_sip.so failed! I had this as well when I tried: Noload = res_musiconhold.co However If you leave it on default: Load = res_musiconhold.so it should work! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to share Zap channels in 2 Asterisk servers
Please reply with sip.conf extension.conf for both servers. Hard to tell what the problem is without see config info - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 22, 2004 11:39 AM Subject: [Asterisk-Users] How to share Zap channels in 2 Asterisk servers Hello I am trying to setup Asterisk on 2 servers PBX300 and PBX200. PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap device. Softphone from PBX200 can talk to softphone on PBX300 but no outgoing call from PBX200. I can call from PBX300 outside but I am unable to configure soft Phone defined in PBX200 to dial out side using PBX300 Zap devices. I am geting error message Rejected connect attempt from PBX200. Please help if this is possible. Thanks Deepak This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RxFAX generates no tiff file
Dunno about not being able to generate a tiff, I got rxfax to do that, but they're badly malformed. http://roanoke-voip01.psknet.com/fax/ -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Heininger Sent: Saturday, May 22, 2004 12:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RxFAX generates no tiff file Hi, I am trying to receive a fax with the spandsp library. The sending fax says success but there is no tiff file generated. I use exten = 7000,1,rxfax(/tmp/testfax.tif) in my extensions.conf. The connection is via SIP/G.711 as I have read on the list that this can sometimes work (I know Fax over IP is troublesome without T.38). I think the transmission should not be the problem because of the success on the sending fax. This is the debug output. Am I missing something? TIA, Mike *CLI-- Executing RxFAX(SIP/uid-c5b6, /tmp/testfax.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up Slow carrier down T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Saturday, May 22, 2004 2:07 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop Leif Madsen wrote: Afternoon all, I'm trying to load Asterisk, however I am getting the following error: [skipping res_musiconhold.so] [chan_sip.so]May 22 18:42:24 WARNING[16384]: loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop May 22 18:42:24 WARNING[16384]: loader.c:421 load_modules: Loading module chan_sip.so failed! I had this as well when I tried: Noload = res_musiconhold.co However If you leave it on default: Load = res_musiconhold.so it should work! Wow, it DID work. Thanks! Leif Madsen. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamic SIP.CONF
Brian Cuthie wrote: So I've been kind of struggling with the notion of making my Asterisk implementations dynamic, too. While I'd like to make everything directly database driven, I'm not sure Asterisk is quite there yet. I've been thinking of writing something that creates appropriate configuration files from the database on a periodic basis, and then does an Asterisk reload. This would introduce a small delay into configuration changes, but it does have other benefits such as decoupling the design of the database from Asterisk. Any thoughts? This is exactly what I do - works very well so far :) I guess that it will reach scalability limits at some stage...but so far, so good... I write out: users-sip.conf users-iax.conf users-voicemail.conf mapping.conf(username- extension) These are #included into the main files. I restart Asterisk via the manager port, since 'asterisk -r -x reload' doesn't return properly the web UI 'sticks' horribly otherwise. I complement this by using ODBCGet in the dialplan. (Previously I #included dnd.conf, calldiversion.conf to achieve this functionality) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop
Leif Madsen wrote: I'm trying to load Asterisk, however I am getting the following error: [skipping res_musiconhold.so] [chan_sip.so]May 22 18:42:24 WARNING[16384]: loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop May 22 18:42:24 WARNING[16384]: loader.c:421 load_modules: Loading module chan_sip.so failed! I've tried doing res_musiconhold.so=no in modules.conf with no change. This res is a requirement for current versions of chan_sip So, definitely *don't* have this in modules.conf: noload = res_musiconhold.so The question therefore is why is this res being skipped? Missing musiconhold.conf ? F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFAX generates no tiff file
Am 22.05.2004 um 20:09 schrieb Troy Settle: Dunno about not being able to generate a tiff, I got rxfax to do that, but they're badly malformed. This is more than I get ;-) Does the fax on the other side get a success message? I get fax-rx-audio and fax-tx-audio files in /tmp but no tiff output file. Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: dial application - continue in context
Hi Philipp On Sat, May 22, 2004 at 02:29:18PM +0200, Philipp von Klitzing wrote: Note: The h extension is not reliable enough to solve your problem. What is the problem with the hangup extension? Thanks in advance -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: dial application - continue in context
Hi! On Sat, May 22, 2004 at 02:29:18PM +0200, Philipp von Klitzing wrote: Note: The h extension is not reliable enough to solve your problem. What is the problem with the hangup extension? Not reliable - ask bkw for details, he can elaborate. P. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sipura and STUN (was: rejected NOTIFY requests)
At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote: [snip] Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk can handled the NAT traversal all by itself with Qualify (as John points out) disabling the NOTIFY will not change anything. The NOTIFY will in no way affect the status - unreachable/reachable. Another problem with the SIPURA is the lack of a working STUN solution. Even Grandstream works better with NAT today. /O Do you have difficulties with the Sipura SPA-2000 (or other Sipura boxes) and Asterisk? I've found no problems, even behind NAT, though I have only tried behind one or two NAT devices (OpenBSD and Apple Airport.) It's surprising that Sipura doesn't include STUN as an option - their list of options is so huge that I always assumed I had just missed it, but now that I look closer I suppose you're right. Do Asterisk users even really need STUN? I've never found it to be required after the NAT issues were worked out of Asterisk... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fwd on busy when calling multiple extensions at once
IMHO, the Polycom IP600 is a superior phone to the cisco 79XX phones. I base this off of having had both an IP600 and a 7960. The two advantages the 7960 had over the IP600 was appearance and ease of configuration. Outside of that, the IP600 (IMHO) beat the cisco hands down. Now, you MAY want to try registering all 6 lines on the polycom to the same line and see if the phone handles that as well as the cisco. If it does, then you are set. Otherwise, you will need some complex configuration work in your extensions.conf to achieve what you are looking to achieve. Some thoughts: What do you want to happen when one of the call takers has all 6 lines in use? Have you considered using queues to do what you need? -Chris On 10:08 AM 5/22/2004, Brian Cuthie wrote: You might consider using the Cisco SIP phones. They're smart enough to accept incoming calls for as many call appearances you have with the same SIP registration. -brian Tor Roberts wrote: Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3 phones ring on the first extension, but the dispatcher who is on a call, her phone does not ring. I want her second extension ring along with the other 3 phones first extensions. In sip.conf I have all the extensions set to incominglimit=1 and the pertinent part of extensions.conf is: exten = s,1,Dial(SIP/5000SIP5001SIP5002SIP5003,20,tr) exten = s,2,Dial(SIP/5004SIP5005SIP5006SIP5007,20,tr) and so on. If anybody has any insight, or a better solution, that would be great. Thanks, -Tor Roberts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Sipura and STUN (was: rejected NOTIFY re quests)
Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay -Original Message- From: John Todd [mailto:[EMAIL PROTECTED] Sent: Saturday, May 22, 2004 1:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Sipura and STUN (was: rejected NOTIFY requests) At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote: [snip] Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk can handled the NAT traversal all by itself with Qualify (as John points out) disabling the NOTIFY will not change anything. The NOTIFY will in no way affect the status - unreachable/reachable. Another problem with the SIPURA is the lack of a working STUN solution. Even Grandstream works better with NAT today. /O Do you have difficulties with the Sipura SPA-2000 (or other Sipura boxes) and Asterisk? I've found no problems, even behind NAT, though I have only tried behind one or two NAT devices (OpenBSD and Apple Airport.) It's surprising that Sipura doesn't include STUN as an option - their list of options is so huge that I always assumed I had just missed it, but now that I look closer I suppose you're right. Do Asterisk users even really need STUN? I've never found it to be required after the NAT issues were worked out of Asterisk... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Dynamic SIP.CONF
On Sat, May 22, 2004 at 05:39:48PM +0100, Fran Boon wrote: Only by extending the functionality of sip friends to include this extra field... In chan_sip.c the configuration data from sip.conf is used to build a list of sip friends. Checks for waiting voice mail are done for the members of this list one by one. If the configuration data is selected from the mysql table, it is never added to this list. Some temporary structure is used to store this data. It won't be sufficient to store the required data in the mysql table an transfer it to the internal structure. IMHO it is preferable to write some file included in sip.conf and to do a sip reload if necessary. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sipura and STUN (was: rejected NOTIFY re quests)
Beyond this, you can still just use the NAT keepalive in the Sipura. While It only provides for either a NOTIFY or REGISTER (which both generate errors in asterisk) if you change it to something else (I just have it send blank, but a few ... or anything will do) asterisk won't complain and the data is sent every few seconds, keeping the firewall open. I've also found setting the register to something low (I use 300s) also helps when you do have to use qualify, in case asterisk loses the connection the device will only be offline until the next register. -Steve On May 22, 2004, at 3:32 PM, Darren Nay wrote: Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay -Original Message- From: John Todd [mailto:[EMAIL PROTECTED] Sent: Saturday, May 22, 2004 1:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Sipura and STUN (was: rejected NOTIFY requests) At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote: [snip] Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk can handled the NAT traversal all by itself with Qualify (as John points out) disabling the NOTIFY will not change anything. The NOTIFY will in no way affect the status - unreachable/reachable. Another problem with the SIPURA is the lack of a working STUN solution. Even Grandstream works better with NAT today. /O Do you have difficulties with the Sipura SPA-2000 (or other Sipura boxes) and Asterisk? I've found no problems, even behind NAT, though I have only tried behind one or two NAT devices (OpenBSD and Apple Airport.) It's surprising that Sipura doesn't include STUN as an option - their list of options is so huge that I always assumed I had just missed it, but now that I look closer I suppose you're right. Do Asterisk users even really need STUN? I've never found it to be required after the NAT issues were worked out of Asterisk... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P problems with 1 FXS, 1 FXO
David, Not sure if you already got a reply or not - but it looks to me like your FXO module is on port 3 - not 2 (see the dmesg output). Give that a try. HTH- Ben On Wednesday, May 19, 2004, at 12:51 PM, David Creemer wrote: Hi- I'm totally stumped configuring my TDM400P with one FXS and one FXO module. Before I got the FXO module, I used to have an X101P, and everything was working very well. Now * doesn't seem to recognize the FXO channel. I've searched the wiki and the list archives. Stock Debian 3.0 stable installation. Any advice? Thanks. -- David Here's my configuration: modprobe zaptel modprobe wcfxs report no errors. box:/etc/asterisk# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. --- So it looks like things are OK so far. Here's the relevant portion of my zaptel.conf: defaultzone=us # load FXO X100P channel 1, kewlstart signalling # turned off, card removed #fxsks=1 # load FXS TDM400P channel 1, kewlstart signalling fxoks=1 # load FXO TDM400P channel 2, kewlstart signalling fxsks=2 And here's what dmesg reports: Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 12 for device 00:09.0 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO FXS Module 1: Not installed Module 2: Installed -- AUTO FXO Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 0 (United States / North America) --- the relevant portions of my zapata.conf are: [channels] language=en usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes busydetect=yes callprogress=yes ; interfaces for internal analog phones signalling=fxo_ks threewaycalling=yes transfer=yes group=1 context=from-internal callerid=Creemer 01 channel = 1 mailbox=01 ; interfaces to the external PSTN line signalling=fxs_ks context=from-pstn group=2 channel = 2 --- starting asterisk gives: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found May 19 10:42:20 DEBUG[1024]: chan_zap.c:1077 update_conf: Updated conferencing on 1, with 0 conference users -- Registered channel 1, FXO Kewlstart signalling May 19 10:42:20 WARNING[1024]: chan_zap.c:665 zt_open: Unable to specify channel 2: No such device May 19 10:42:20 ERROR[1024]: chan_zap.c:5340 mkintf: Unable to open channel 2: No such device here = 0, tmp-channel = 2, channel = 2 May 19 10:42:20 ERROR[1024]: chan_zap.c:7376 setup_zap: Unable to register channel '2' May 19 10:42:20 WARNING[1024]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 -- Unregistered channel 2 May 19 10:42:20 WARNING[1024]: loader.c:408 load_modules: Loading module chan_zap.so failed! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rejected NOTIFY requests
Another problem with the SIPURA is the lack of a working STUN solution. Even Grandstream works better with NAT today. /O I disagree. We have hundreds of Sipura customers using STUN with our SER Solution. The are the most stable SIP UA we have ever tested. We had to dump loads of Grandstream phones on Ebay due to their unstable STUN operation. -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Failure while compiling
Hi guys! I just try to compile Asterisk with make all and get the following lines multible times: cli.c:31:19: build.h: No such file or directory dlfcn.c:40:25: mach-o/dyld.h: No such file or directory dlfcn.c:41:26: mach-o/nlist.h: No such file or directory dlfcn.c:42:28: mach-o/getsect.h: No such file or directory Can someone tell me what's exactly missing? Regards from Munich Julian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Failure while compiling
Nothing, it's normal to get those errors - I get them all the times I compile asterisk on Linux, FreeBSD, and Windows. Your failure to compile is being caused elsewhere. - Joshua Colp. - Original Message - From: Julian Pawlowski [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 22, 2004 5:44 PM Subject: [Asterisk-Users] Failure while compiling Hi guys! I just try to compile Asterisk with make all and get the following lines multible times: cli.c:31:19: build.h: No such file or directory dlfcn.c:40:25: mach-o/dyld.h: No such file or directory dlfcn.c:41:26: mach-o/nlist.h: No such file or directory dlfcn.c:42:28: mach-o/getsect.h: No such file or directory Can someone tell me what's exactly missing? Regards from Munich Julian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Failure while compiling
Nothing, it's normal to get those errors - I get them all the times I compile asterisk on Linux, FreeBSD, and Windows. Your failure to compile is being caused elsewhere. Ah okay, thanks. Although make all is successfully, I say these messages and tought that it could result in some incorrect behavior of asterisk anywhere. Regards, Julian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sipura and STUN (was: rejected NOTIFY requests)
John Todd wrote: At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote: [snip] Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk can handled the NAT traversal all by itself with Qualify (as John points out) disabling the NOTIFY will not change anything. The NOTIFY will in no way affect the status - unreachable/reachable. Another problem with the SIPURA is the lack of a working STUN solution. Even Grandstream works better with NAT today. /O Do you have difficulties with the Sipura SPA-2000 (or other Sipura boxes) and Asterisk? I've found no problems, even behind NAT, though I have only tried behind one or two NAT devices (OpenBSD and Apple Airport.) It's surprising that Sipura doesn't include STUN as an option - their list of options is so huge that I always assumed I had just missed it, but now that I look closer I suppose you're right. Do Asterisk users even really need STUN? I've never found it to be required after the NAT issues were worked out of Asterisk... No, no problems with a Sipura and Asterisk. The Sipura is impressive, so I'm surprised that it doesn't support STUN and NAT in a good way, so we could enable canreinvite= /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rejected NOTIFY requests
Andres wrote: Another problem with the SIPURA is the lack of a working STUN solution. Even Grandstream works better with NAT today. /O I disagree. We have hundreds of Sipura customers using STUN with our SER Solution. The are the most stable SIP UA we have ever tested. We had to dump loads of Grandstream phones on Ebay due to their unstable STUN operation. Great. I need to upgrade and test again. I really need STUN and DNS SRV to work as expected. Thank you Andres! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My TDM-400P FXO experience
I see that * refers to the channels this way on the console output, but I get warnings when I try to use the new naming in the extensions.conf dial plan - anyone else notice this? How do you refer to the channels in extensions? On Tuesday, May 18, 2004, at 07:50 PM, Leo Ann Boon wrote: f. Be careful about the zap channel naming. With the old XP101, the first channel (card) is Zap/1 and the second Zap/2. With the TDM, it's Zap/1-1, Zap/2-1 ... Zap/4-1 for the 4 ports on the first card and Zap/1-2 ... Zap/4-2 for the second card. You might need to update your dial plan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fwd on busy when calling multiple extensions at once
Tor Roberts wrote: Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3 phones ring on the first extension, but the dispatcher who is on a call, her phone does not ring. I want her second extension ring along with the other 3 phones first extensions. In sip.conf I have all the extensions set to incominglimit=1 and the pertinent part of extensions.conf is: exten = s,1,Dial(SIP/5000SIP5001SIP5002SIP5003,20,tr) exten = s,2,Dial(SIP/5004SIP5005SIP5006SIP5007,20,tr) and so on. If anybody has any insight, or a better solution, that would be great. Thanks, -Tor Roberts OK. The following assumes that your phones have 4 phones with 2 lines a piece. Phone 1: 5000, 5004 Phone 2: 5001, 5005 Phone 3, 5002, 5006 Phone 4, 5003, 5007 Adapt it as you choose. It will ring all four phones at a time. If line 1 of a phone is busy, it will ring line 2 of that phone. This is off the top of my head and hadn't been tested on my asterisk server but, I'm pretty sure it will work. I have a few lines set up in this manner. [extensions] exten = s,1,Dial(LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED],20,tr) exten = s,2,hangup exten = 5000,1,Dial(SIP/5000) exten = 5000,2,hangup exten = 5000,102,Dial(SIP/5004) exten = 5000,103,hangup exten = 5001,1,Dial(SIP/5001) exten = 5001,2,hangup exten = 5001,102,Dial(SIP/5005) exten = 5001,103,hangup exten = 5002,1,Dial(SIP/5002) exten = 5002,2,hangup exten = 5002,102,Dial(SIP/5006) exten = 5002,103,hangup exten = 5003,1,Dial(SIP/5003) exten = 5003,2,hangup exten = 5003,102,Dial(SIP/5007) exten = 5003,103,hangup John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HOW do I restore voicemail from backups?
I am trying to recreate an * server from backups. I copied /var/spool/asterisk/voicemail/context/109/INBOX/* from backups. The voicemail files got restored msg.gsm msg.txt msg.wav but when the user goes into voicemail, * says there is no voicemail. Thanks! Paul Paul Mahler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.signate.com/ Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse SIP
Andres wrote: [EMAIL PROTECTED] wrote: Which providers give you a jitter buffer? In Europe: VoipTalk and Magrathea. In the US: Iconnecthere. I am sure there are more. Clearpath gives jitter buffer as well. http://www.clearpath1.com/ John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk firewall config
The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension numberpassword could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there isn't one documented... Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller ID with BT CD50
Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? So, that leaves me with the modem route, which seems more and more unlikely, due to the seeming difficulties finding a modem that will *definitely* do it, or the CD50 mod. Which brings me to my question (finally) Has anyone done this [the CD50] mod? It seems the CD50 can be found for a few quid, and I'm not afraid of my soldering iron... I just wondered how people in the UK were capturing callerid. there is so much more you can get asterisk to do if you have access to callerid info. By the way. much as I'd like to do this by switching to ISDN, and be done with it, this server is at my home for me to play with, and ISDN is *not* cheap in fact it would roughly double my quarterly phone bill. other than the price, ISDN would be my solution of choice! Thanks in advance, Karl This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk firewall config
I personally only allow IAX2 in and out from my asterisk box, due to the simplicity of one (udp) port. I do not relish the thought of trying to open the port ranges for SIP securely! As long as your inbound stuff in iax.conf lands in a sensible context, inbound connections would only be able to call your internal extensions, and not make cost calls. Hope that helps Karl -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tony Hoyle Sent: 22 May 2004 23:11 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk firewall config The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension numberpassword could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there isn't one documented... Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ast_rtp_read: Unknown RTP codec 72 received
Hi, i'd like to know more about this issue, i'm always getting this message while in call with anyone from sip to zap or zap to sip. ast_rtp_read: Unknown RTP codec 72 received here is my current setup: client side, x-lite, with the transmit silence to yes, using ulaw,alaw on asterisk server side: sip.conf contain allow=ulaw and allow=alaw dtmfmode=inband So i always get this anoying notice and i cannot find any doc about fixing it. I have try to put rfc2833 or info for dtmfmode, still giving this result. Plus for the one on the zap side (using a regular phone) , he is hearing me like crystal. Me, using a sip x-lite phone software, i am always hearing parasite. Thank in advance, i would really appreciate some help about this issue. Here is my email for the one who know some of the answer : [EMAIL PROTECTED] Sincerely JF ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My TDM-400P FXO experience
Hi On Tuesday, May 18, 2004, at 07:50 PM, Leo Ann Boon wrote: f. Be careful about the zap channel naming. With the old XP101, the first channel (card) is Zap/1 and the second Zap/2. With the TDM, it's Zap/1-1, Zap/2-1 ... Zap/4-1 for the 4 ports on the first card and Zap/1-2 ... Zap/4-2 for the second card. You might need to update your dial plan. that sounds very strange are you sure? as far as I know each Zap channel is unique, so with 2 cards you should have from Zap/1 to Zap/8 The difference between Zap/1-1 and Zap/1-2 is (for example) when you have 2 calls on the same zap channel, ie when you have a call on the phone on Zap/1-1 and pressing the flash key, you create Zap/1-2 on which you can dial another exten. I don't think that Zap/1-1 and Zap/1-2 are first channels on different cards at all please double check that (as I'll do...) Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_queue and app_groupcount
The new app_groupcount looks great for most applications but it a is a step back for call queueing... since app_queue calls physical interfaces and not extensions, app_groupcont can't be used to limit the calls passed to a dynamically added agent. I presently use the broken sip incominglimit feature (even though it's less than ideal as it also limits outgoing calls preventing consultative transfer using sip refer commands) I could start to use the agents app with agentcallbacklogin to (almost) emulate the current behaviour and use app_groupcount - I can automate the login using agentcallback login, but not the logoff, it prompts for an extension to forward to requireing # to pressed to log off - is there any way round this? I'd prefer to keep the simplicity of simply dialing one number to log on in or out of the queue from any phone, without having to define agentids, passwords, etc which we don't need. I hope incominglimit and outgoing limit aren't going to be removed entirely... -- Julien ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fwd on busy when calling multiple extensions at once
Chris, As far as the Cisco phones, they are not an option as I already have the Polycoms. The Ciscos are overpriced anyway. It was my understanding that asterisk would not let you register the same extension more than once. If that is not the case, I will try to register the same extension to all 6 lines. If all 6 lines are used on any of the phones then I imagine that only the other 3 phones will ring. I don't think this will happen, as I only have 8 incoming lines. If it does become a problem, then I could enable call waiting on the last 2 lines so that each phone can handle 8 calls. Thank you for your advice! -Tor Roberts Chris A. Icide wrote: IMHO, the Polycom IP600 is a superior phone to the cisco 79XX phones. I base this off of having had both an IP600 and a 7960. The two advantages the 7960 had over the IP600 was appearance and ease of configuration. Outside of that, the IP600 (IMHO) beat the cisco hands down. Now, you MAY want to try registering all 6 lines on the polycom to the same line and see if the phone handles that as well as the cisco. If it does, then you are set. Otherwise, you will need some complex configuration work in your extensions.conf to achieve what you are looking to achieve. Some thoughts: What do you want to happen when one of the call takers has all 6 lines in use? Have you considered using queues to do what you need? -Chris On 10:08 AM 5/22/2004, Brian Cuthie wrote: You might consider using the Cisco SIP phones. They're smart enough to accept incoming calls for as many call appearances you have with the same SIP registration. -brian Tor Roberts wrote: Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3 phones ring on the first extension, but the dispatcher who is on a call, her phone does not ring. I want her second extension ring along with the other 3 phones first extensions. In sip.conf I have all the extensions set to incominglimit=1 and the pertinent part of extensions.conf is: exten = s,1,Dial(SIP/5000SIP5001SIP5002SIP5003,20,tr) exten = s,2,Dial(SIP/5004SIP5005SIP5006SIP5007,20,tr) and so on. If anybody has any insight, or a better solution, that would be great. Thanks, -Tor Roberts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk firewall config
The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension numberpassword could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there isn't one documented... Tony, What you open up (and how you restrict access) is really a function of the resources you have available. Example, on some firewalls you can open a ton of ports, but then limit which IP's can actually use them. I think there is a permit= statement for sip def's that limit which IP's can use that sip definition. If that's not enough, implement IP tables as another mechanism to restrict access. All depends on what you've got available. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk firewall config
Hi Il dom, 2004-05-23 alle 00:11, Tony Hoyle ha scritto: The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? all depends on what you need to do. if you use only zap channels and no Voip, perhaps the only port you need to open is ssh (if using it, of course) if you plan to do only IAX, only port 4569 UDP needs to be opened. but if you plan to do only sip you need only port 5060 UDP and 1 to 2 UDP for sip rtp stream (configurable into rtp.conf) so... all depends :) I'm concerned as anyone that could guess an extension numberpassword could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there isn't one documented... mmmh... setting into the extension seems to me the same as setting into iax.conf (or sip.conf), or not? otherwise... use very strange passwords along with superstrange usernames I bet someone to get a login data like username : 2h729872pcnt with pw : inr2.f2f2232DDFW3r or not :) ? -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
Karl Dyson wrote: Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? From the details on http://www.ainslie.org.uk/callerid/cli_faq.htm it sounds like it wouldn't be too hard to implement, however: The only manufacturers that have ever supported BT Caller ID are Pace, Hayes (Europe), and 3Com/US Robotics. It then goes on to state all 3 of those manufactures no longer support it. I wonder if the low cost geographic VOIP numbers support it? Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip call using name in sip.conf
i try to place a call exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]:5061,60,Ttr) where sip.conf has an entry [foo] secret=torture callerid=local ext 103 1914666 type=friend fromuser=asterisk auth=both host=dynamic canreinvite=yes context=in-914 mailbox=001 i get May 22 23:11:31 WARNING[140400128]: chan_sip.c:902 create_addr: \ No such host: foo May 22 23:11:31 NOTICE[140400128]: app_dial.c:536 dial_exec: \ Unable to create channel of type 'SIP' the sip service is registered foo/foo 209.20.186.194 (D) 255.255.255.255 5060 Unmonitored and i get the same result if it is not dynamic foo/foo 209.20.186.194 255.255.255.255 5061 Unmonitored clues appreciated randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip call using name in sip.conf
i try to place a call exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]:5061,60,Ttr) ^^^ That foo name needs to be changed to an IP address of whatever it is that is suppose to handle the call. Asterisk is doing a DNS name lookup and can't resolve it, therefore no such host. Also, not sure what :5061 is suppose to represent in your example. where sip.conf has an entry [foo] secret=torture callerid=local ext 103 1914666 type=friend fromuser=asterisk auth=both host=dynamic canreinvite=yes context=in-914 mailbox=001 i get May 22 23:11:31 WARNING[140400128]: chan_sip.c:902 create_addr: \ No such host: foo May 22 23:11:31 NOTICE[140400128]: app_dial.c:536 dial_exec: \ Unable to create channel of type 'SIP' the sip service is registered foo/foo 209.20.186.194 (D) 255.255.255.255 5060 Unmonitored and i get the same result if it is not dynamic foo/foo 209.20.186.194 255.255.255.255 5061 Unmonitored clues appreciated randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fwd on busy when calling multiple extensions at once
Tor Roberts wrote: It was my understanding that asterisk would not let you register the same extension more than once. If that is not the case, I will try to register the same extension to all 6 lines. On the 7960's, * does not get upset with having multiple appearances of the same line on a 7960. You just config the phone the same way with all six lines. If all 6 lines are used on any of the phones then I imagine that only the other 3 phones will ring. That's how it works. One gotcha to watch for though is if you have a +101 priority that goes to voicemail or something. What would happen is that when the VM answers, the other phones will stop ringing. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip call using name in sip.conf
Randy Bush wrote: i try to place a call exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]:5061,60,Ttr) where sip.conf has an entry [foo] secret=torture callerid=local ext 103 1914666 type=friend fromuser=asterisk auth=both host=dynamic canreinvite=yes context=in-914 mailbox=001 Randy, Try the following: exten = _X.,1,Dial(SIP/foo:5061,60,Ttr) This will cause asterisk to send the call to sip peer foo. If you're trying to send the call to a specific extension on host foo.bar, you'll need to do something like this: exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]:5061,60,Ttr) If the other side is an * box as well, I highly recommend you use IAX2 and not SIP. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk firewall config
Brancaleoni Matteo wrote: if you plan to do only IAX, only port 4569 UDP needs to be opened. but if you plan to do only sip you need only port 5060 UDP and 1 to 2 UDP for sip rtp stream (configurable into rtp.conf) so... all depends :) Surely it depends on who's calling me - if they're using a SIP phone it'll come in over the SIP port, and if they're using an IAX phone it'll come in over the IAX port - ie there's this context in the default iax.conf: [guest] type=user context=default callerid=Guest IAX User Which I assume is there for a reason... otherwise why have it? btw. how many rtp streams do I need? I only have 1 phone at the moment (max. will be about 4 I think). otherwise... use very strange passwords along with superstrange usernames I bet someone to get a login data like username : 2h729872pcnt with pw : inr2.f2f2232DDFW3r I already use pretty strange/long passwords... the recommendation always seems to be make username==extension number, though. Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk slashdotted
Congradulations to the Asterisk gang on getting slashdotted! http://slashdot.org/article.pl?sid=04/05/22/1840220 Cheers, Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip call using name in sip.conf
Randy == Randy Bush [EMAIL PROTECTED] writes: Randy i try to place a call Randy exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]:5061,60,Ttr) Randy where sip.conf has an entry Randy [foo] Randy type=friend I do not beleive that will work for type=friend. If you use separate type=peer and type=user blocks in sip.conf it may work. Expecially if you also specify a port in the Dial(). Else, use the hostname (or a const). -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] e164.org
So I just saw this VoIP-centric article at slashdot (http://slashdot.org/article.pl?sid=04/05/22/1840220) which mentions e164.org. It's a non-profit public DNS root designed to map phone numbers to Internet protocols. Is anyone on this list actually using this? They have asterisk config instructions: http://www.e164.org/config.php I wonder if someone can help me understand this. Let's say I configure my asterisk box to use e164 and then I try to call a phone number in Germany. I'm in the U.S.A. So if the number I'm calling in Germany is registered in e164's dns, would my call be routed directly via their voip provider? Or directly to their asterisk box? And would it be free? If that's the case, it sounds kind of cool, but probably won't be much use until lots of people sign up. Any explanation appreciated. Thanks. Simon in New Orleans P.S.- Yes, I did read their FAQ. http://wiki.e164.org/moin.cgi/FrequentlyAskedQuestions ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] e164.org
Simon Dorfman wrote: I wonder if someone can help me understand this. Let's say I configure my asterisk box to use e164 and then I try to call a phone number in Germany. I'm in the U.S.A. So if the number I'm calling in Germany is registered in e164's dns, would my call be routed directly via their voip provider? Or directly to their asterisk box? And would it be free? From the looks of it, they're just a directory... it looks like their not running asterisk themselves. They use something called EnumLookup which I guess is some kind of plugin/script. If the number you're calling is in their database, it calls the VOIP number directly, otherwise it calls the POTS number It's an interesting idea. Of course having a huge database of names/addresses/phone numbers can be quite lucrative too. Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] e164.org
On Sat, 2004-05-22 at 18:08, Tony Hoyle wrote:
Simon Dorfman wrote:
I wonder if someone can help me understand this. Let's say I configure my
asterisk box to use e164 and then I try to call a phone number in Germany.
I'm in the U.S.A. So if the number I'm calling in Germany is registered in
e164's dns, would my call be routed directly via their voip provider? Or
directly to their asterisk box? And would it be free?
From the looks of it, they're just a directory... it looks like their not
running asterisk themselves.
It's a DNS root, that Asterisk (via the EnumLookup application) can
use. The EnumLookup() application will resolve the number to a dial()
channel.
ie:
; north america enum
exten = _1NX,1,Playback(doing-enum-lookup)
exten = _1NX,2,EnumLookup(${EXTEN})
exten = _1NX,3,BackGround(enum-lookup-successful)
exten = _1NX,4,Dial(${ENUM},30,tr)
exten = _1NX,5,Hangup
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup
To get * to resolve against e164.org, add:
search = e164.org
to /etc/asterisk/enum.conf.
So yes Simon, if you called someone in Germany and it was the zone, your
call would be switched over the 'net. If not, you could drop it to
NuFone or some other carrier.
They use something called EnumLookup which I guess is some kind of
plugin/script. If the number you're calling is in their database, it calls
the VOIP number directly, otherwise it calls the POTS number
Or whatever else your dial plan wants to do.
Matthew
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[Asterisk-Users] Asterisk-oh323 0.6.1 Compiling problem
Hi, i'm having another problem I can't work out - make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make: *** No rule to make target `ccflags'. Stop. make: *** No rule to make target `ccflags'. Stop. make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.1/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 g++ -Wall -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.5.2\ -DOPENH323VERSION=\1.12.2\ -I/usr/include/openssl -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -I/usr/src/openh323/include/openh323 -I../asterisk-driver -x c++ -Os -g -c asteriskaudio.cxx -o asteriskaudio.o asteriskaudio.cxx: In method `PAsteriskSoundChannel::~PAsteriskSoundChannel ()': asteriskaudio.cxx:163: `baseChannel' undeclared (first use this function) asteriskaudio.cxx:163: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: *** [asteriskaudio.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.1/wrapper' make: *** [subdirs_all] Error 1 I don't know whats going wrong here, I think I have all the libraries installed. Asterisk runs fine and is CVS version so what gives? Thanks? Nicholas Ruddick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] e164.org
Dean Collins wrote: Tony, as per you inference that e164 are up to something shady, you should talk to one of the founders Duane, he currently has about 5 open If it's the same duane who runs cacert he probably means well... however having read the site I'm still not sure whether i'd use it myself (it means trusting an external database to produce a least cost route.. I'm just not that trusting). Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-oh323 0.6.1 Compiling problem
Check your README file again. In order to compile 0.6.1 you need newer versions of pwlib and openh323 (1.6.6 and 1.13.5) Then it should work just fine Pablo -- Pablo Endres [EMAIL PROTECTED] ComVoz Comunications ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] e164.org
Matthew Asham wrote:
; north america enum
exten = _1NX,1,Playback(doing-enum-lookup)
exten = _1NX,2,EnumLookup(${EXTEN})
exten = _1NX,3,BackGround(enum-lookup-successful)
exten = _1NX,4,Dial(${ENUM},30,tr)
exten = _1NX,5,Hangup
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup
Interesting.. how does it know to go to '6', or does it just jump +4
on failure?
That reminds me I seriously need to restructure my extensions.conf... there's
no way currently I could add anything like that without major surgery (only
discovered the 'local' target this afternoon so I have everything copied/pasted).
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
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Re: [Asterisk-Users] e164.org
You know, sleep deprivation cause people to do dumb things. The example
I pasted was hastily pasted and renumbered,
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup
are actually:
exten = _1NX,103,Playback(enum-lookup-failed)
exten = _1NX,104,Hangup
Duane wrote up some more detailed examples at
http://www.e164.org/config.php.
Sorry for not proofing that when I posted it. I'll go sleep now.
On Sat, 2004-05-22 at 18:46, Tony Hoyle wrote:
Matthew Asham wrote:
; north america enum
exten = _1NX,1,Playback(doing-enum-lookup)
exten = _1NX,2,EnumLookup(${EXTEN})
exten = _1NX,3,BackGround(enum-lookup-successful)
exten = _1NX,4,Dial(${ENUM},30,tr)
exten = _1NX,5,Hangup
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup
Interesting.. how does it know to go to '6', or does it just jump +4
on failure?
That reminds me I seriously need to restructure my extensions.conf... there's
no way currently I could add anything like that without major surgery (only
discovered the 'local' target this afternoon so I have everything copied/pasted).
Tony
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[Asterisk-Users] T100P HDLC configuration
Thank you, Michael
I tried to switch to FR mode... but it did not help. I tied DLCI as 16
and 99... the same result.
I attached one more full config from Netopia and from my Linux+Zaptel
T100P systems.
DEVICE=hdlc0
# MODE=hdlc
# MODE=cisco
MODE=fr
NETMASK=255.255.255.252
GATEWAY=REMOTE_IPADDR
# FR
FR_LMI=ansi
FR_PVC=pvc0
FR_DLC=16
# FR_DLC=99
case "$1" in
'start')
echo "Loading T1/HDLC modules..."
/sbin/modprobe zaptel
/sbin/modprobe wct1xxp
/sbin/modprobe hdlc
/sbin/modprobe syncppp
/sbin/ztcfg -vvv
echo -n "Configuring HDLC interfaces, with mode
\"${MODE}\""
if [ "${MODE}" == "hdlc" -o "${MODE}" == "cisco" ]; then
echo "..."
/sbin/sethdlc ${DEVICE} ${MODE}
/sbin/ifconfig ${DEVICE} ${LOCAL_IPADDR}
pointopoint ${REMOTE_IPADDR}
/sbin/route add -net ${NETWORK} netmask
${NETMASK} ${DEVICE}
echo "Configuring default gateway..."
/sbin/route add default gw ${GATEWAY} metric 1
${DEVICE}
elif [ "${MODE}" == "fr" ]; then
echo ", LMI \"${FR_LMI}\"..."
/sbin/sethdlc ${DEVICE} ${MODE} lmi ${FR_LMI}
/sbin/sethdlc ${DEVICE} create ${FR_DLC}
/sbin/ifconfig ${DEVICE} up
echo "Configuring Frame-Relay PVC
\"${FR_PVC}\"..."
/sbin/ifconfig ${FR_PVC} ${LOCAL_IPADDR}
pointopoint ${REMOTE_IPADDR}
/sbin/route add -net ${NETWORK} netmask
${NETMASK} ${FR_PVC}
echo "Configuring default gateway..."
/sbin/route add default gw ${GATEWAY} metric 1
${FR_PVC}
else
echo ", unknown mode..."
fi
;;
'stop')
echo "Unloading default gateway..."
/sbin/route del default
echo -n "Unloading HDLC configuration."
if [ "${MODE}" == "hdlc" -o "${MODE}" == "cisco" ]; then
echo ", hdlc/cisco mode..."
/sbin/route del -net ${NETWORK} netmask
${NETMASK} ${DEVICE}
/sbin/ifconfig ${DEVICE} down
elif [ "${MODE}" == "fr" ]; then
echo ", frame-relay mode..."
/sbin/route del -net ${NETWORK} netmask
${NETMASK} ${FR_PVC}
/sbin/ifconfig ${FR_PVC} down
/sbin/sethdlc ${DEVICE} delete
/sbin/ifconfig ${DEVICE} down
else
echo ", unknown mode..."
fi
echo "Unloading T1/HDLI modules..."
rmmod wct1xxp zaptel hdlc syncppp
;;
'restart')
$0 stop
sleep 1
$0 start
;;
*)
echo "usage $0 start|stop|restart"
esac
# $ show config
# frame-relay lmi type ansi
# frame-relay tim none
# hardware acceleration enable yes
# ip gateway REMOTE_IPADDR
# ip route 0.0.0.0/0 REMOTE_IPADDR low
# interface t1 1 buildout 0-0.6
# interface t1 1 channels count 24 start 1 contiguous rate 64k
# interface t1 1 clock source network
# interface t1 1 diagnostic mode normal
# interface t1 1 dle hdlc
# interface t1 1 encoding b8zs
# interface t1 1 framing esf
# interface t1 1 operation mode hdlc
# interface t1 1 priority-queuing enable yes
# interface t1 1 pvc 1 yes
# interface t1 1 pvc 1 enable yes
# interface t1 1 pvc 1 tag "Circuit 1"
# interface t1 1 pvc 1 vpi 0
# interface t1 1 pvc 1 vci 35
# interface t1 1 pvc 1 cp default
# interface t1 1 pvc 1 voice no
# interface t1 1 pvc 1 pcr 0
# interface t1 1 prm-enable no
# interface t1 1 rfc1973 enable no
# interface t1 1 rfc1973 dlci 16
# interface t1 1 rfc1973 lmi none
# interface t1 1 cell-format scrambled
# interface t1 1 unused cell-format idle
# interface t1 1 ds0-autodetect no
# cp 1 yes
# cp 1 tag 36.HCGA.101976.VA
# cp 1 enable yes
# cp 1 dle hdlc
# cp 1 ip enable yes
# cp 1 ip address local LOCAL_IPADDR/30
# cp 1 ip address remote REMOTE_IPADDR/30
# cp 1 ip addressing numbered
# cp 1 ip dhcp client mode standard
# cp 1 ip mask local 255.255.255.252
# cp 1 ip mask remote 255.255.255.252
# cp 1 ip nat enable no
# cp 1 ip nat map-list "Easy-PAT List"
# cp 1 ip nat server-list Easy-Servers
# cp 1 ip negotiate-lan no
# cp 1 ip netbios proxy enable no
# cp 1 ip rip receive both
# cp 1 ip rip transmit no
# cp 1 ip multicast-fwd yes
# cp 1 interface-group primary
# ;Netopia 4622
# name ""
# preferences changes immediate yes
# preferences console default menu
# preferences date format mm/dd/yy
# preferences output format verbose
# preferences output mask bits
# preferences time format 24-hour
#
#
=
--
Thanks and regards,
Vasyl Rublyov
Re: [Asterisk-Users] Asterisk-oh323 0.6.1 Compiling problem
ok done, but now i'm getting different errors -
/usr/src/pwlib/include/ptlib/args.h:389: virtual outside class declaration
/usr/src/pwlib/include/ptlib/args.h:389: non-member function
`UnknownOption (...)' cannot have `const'
method qualifier
/usr/src/pwlib/include/ptlib/args.h:397: parse error before `'
/usr/src/pwlib/include/ptlib/args.h:398: virtual outside class declaration
/usr/src/pwlib/include/ptlib/args.h:398: non-member function
`MissingArgument (...)' cannot have
`const' method qualifier
/usr/src/pwlib/include/ptlib/args.h:401: parse error before `protected'
/usr/src/pwlib/include/ptlib/args.h:405: syntax error before `;'
/usr/src/pwlib/include/ptlib/args.h:407: syntax error before `;'
/usr/src/pwlib/include/ptlib/args.h:409: syntax error before `;'
/usr/src/pwlib/include/ptlib/args.h:411: syntax error before `;'
/usr/src/pwlib/include/ptlib/args.h:413: syntax error before `;'
/usr/src/pwlib/include/ptlib/args.h:417: parse error before `private'
/usr/src/pwlib/include/ptlib/args.h:419: non-member function
`GetOptionCountByIndex (int)' cannot have
`const' method qualifier
/usr/src/pwlib/include/ptlib/args.h:420: syntax error before `('
/usr/src/pwlib/include/ptlib/args.h:428: base class `PArgList' has
incomplete type
/usr/src/pwlib/include/ptlib/args.h:429: ISO C++ forbids declaration of
`PCLASSINFO' with no type
/usr/src/pwlib/include/ptlib/args.h:454: parse error before `'
/usr/src/pwlib/include/ptlib/args.h:465: ISO C++ forbids declaration of
`PString' with no type
/usr/src/pwlib/include/ptlib/args.h:465: `PString' declared as a
`virtual' field
/usr/src/pwlib/include/ptlib/args.h:465: parse error before `('
/usr/src/pwlib/include/ptlib/args.h:470: ISO C++ forbids declaration of
`PString' with no type
/usr/src/pwlib/include/ptlib/args.h:470: `PString' declared as a
`virtual' field
/usr/src/pwlib/include/ptlib/args.h:470: declaration of `int
PConfigArgs::PString'
/usr/src/pwlib/include/ptlib/args.h:465: conflicts with previous
declaration `int PConfigArgs::PString'
/usr/src/pwlib/include/ptlib/args.h:470: parse error before `('
/usr/src/pwlib/include/ptlib/args.h:475: ISO C++ forbids declaration of
`PString' with no type
/usr/src/pwlib/include/ptlib/args.h:475: `PString' declared as a
`virtual' field
/usr/src/pwlib/include/ptlib/args.h:475: declaration of `int
PConfigArgs::PString'
/usr/src/pwlib/include/ptlib/args.h:465: conflicts with previous
declaration `int PConfigArgs::PString'
/usr/src/pwlib/include/ptlib/args.h:475: parse error before `('
/usr/src/pwlib/include/ptlib/args.h:490: parse error before `'
/usr/src/pwlib/include/ptlib/args.h:496: parse error before `'
/usr/src/pwlib/include/ptlib/args.h:501: ISO C++ forbids declaration of
`PString' with no type
/usr/src/pwlib/include/ptlib/args.h:501: declaration of `const int
PConfigArgs::PString'
/usr/src/pwlib/include/ptlib/args.h:465: conflicts with previous
declaration `int PConfigArgs::PString'
/usr/src/pwlib/include/ptlib/args.h:501: parse error before `'
/usr/src/pwlib/include/ptlib/args.h:470: duplicate member
`PConfigArgs::PString'
/usr/src/pwlib/include/ptlib/args.h:475: duplicate member
`PConfigArgs::PString'
/usr/src/pwlib/include/ptlib/args.h:501: duplicate member
`PConfigArgs::PString'
/usr/src/pwlib/include/ptlib/args.h:506: semicolon missing after
declaration of `PConfigArgs'
/usr/src/pwlib/include/ptlib/args.h: In method `void
PConfigArgs::SetSectionName (...)':
/usr/src/pwlib/include/ptlib/args.h:497: `sectionName' undeclared (first
use this function)
/usr/src/pwlib/include/ptlib/args.h:497: (Each undeclared identifier is
reported only once for each
function it appears in.)
/usr/src/pwlib/include/ptlib/args.h:497: `section' undeclared (first use
this function)
/usr/src/pwlib/include/ptlib/args.h: At top level:
/usr/src/pwlib/include/ptlib/args.h:507: parse error before `'
/usr/src/pwlib/include/ptlib/args.h:508: ISO C++ forbids defining types
within return type
/usr/src/pwlib/include/ptlib/args.h:508: two or more data types in
declaration of `SetNegationPrefix'
/usr/src/pwlib/include/ptlib/args.h:508: semicolon missing after
declaration of `class PConfigArgs'
/usr/src/pwlib/include/ptlib/args.h: In function `int SetNegationPrefix
(...)':
/usr/src/pwlib/include/ptlib/args.h:508: `negationPrefix' undeclared
(first use this function)
/usr/src/pwlib/include/ptlib/args.h:508: `prefix' undeclared (first use
this function)
/usr/src/pwlib/include/ptlib/args.h:508: warning: no return statement in
function returning non-void
/usr/src/pwlib/include/ptlib/args.h: At top level:
/usr/src/pwlib/include/ptlib/args.h:513: syntax error before `'
/usr/src/pwlib/include/ptlib/args.h:519: syntax error before `;'
/usr/src/pwlib/include/ptlib/args.h:520: syntax error before `;'
/usr/src/pwlib/include/ptlib/args.h:521: syntax error before `;'
In file included from /usr/src/pwlib/include/ptlib.h:193,
from asteriskaudio.cxx:31:
/usr/src/pwlib/include/ptlib/unix/ptlib/thread.h:150: parse error before `'
Re: [Asterisk-Users] e164.org
You forgot to allow for tel: N+51
bkw
- Original Message -
From: Matthew Asham [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 22, 2004 7:55 PM
Subject: Re: [Asterisk-Users] e164.org
You know, sleep deprivation cause people to do dumb things. The example
I pasted was hastily pasted and renumbered,
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup
are actually:
exten = _1NX,103,Playback(enum-lookup-failed)
exten = _1NX,104,Hangup
Duane wrote up some more detailed examples at
http://www.e164.org/config.php.
Sorry for not proofing that when I posted it. I'll go sleep now.
On Sat, 2004-05-22 at 18:46, Tony Hoyle wrote:
Matthew Asham wrote:
; north america enum
exten = _1NX,1,Playback(doing-enum-lookup)
exten = _1NX,2,EnumLookup(${EXTEN})
exten = _1NX,3,BackGround(enum-lookup-successful)
exten = _1NX,4,Dial(${ENUM},30,tr)
exten = _1NX,5,Hangup
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup
Interesting.. how does it know to go to '6', or does it just jump +4
on failure?
That reminds me I seriously need to restructure my extensions.conf...
there's
no way currently I could add anything like that without major surgery
(only
discovered the 'local' target this afternoon so I have everything
copied/pasted).
Tony
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Re: [Asterisk-Users] RxFAX generates no tiff file
Hi Mike, Your log seems to be incomplete. It stops in the middle of the call. Regards, Steve Mike Heininger wrote: Hi, I am trying to receive a fax with the spandsp library. The sending fax says success but there is no tiff file generated. I use exten = 7000,1,rxfax(/tmp/testfax.tif) in my extensions.conf. The connection is via SIP/G.711 as I have read on the list that this can sometimes work (I know Fax over IP is troublesome without T.38). I think the transmission should not be the problem because of the success on the sending fax. This is the debug output. Am I missing something? TIA, Mike *CLI-- Executing RxFAX(SIP/uid-c5b6, /tmp/testfax.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up Slow carrier down T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] e164.org
'local' target? What's that?
- Original Message -
From: Matthew Asham [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 22, 2004 9:55 PM
Subject: Re: [Asterisk-Users] e164.org
You know, sleep deprivation cause people to do dumb things. The example
I pasted was hastily pasted and renumbered,
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup
are actually:
exten = _1NX,103,Playback(enum-lookup-failed)
exten = _1NX,104,Hangup
Duane wrote up some more detailed examples at
http://www.e164.org/config.php.
Sorry for not proofing that when I posted it. I'll go sleep now.
On Sat, 2004-05-22 at 18:46, Tony Hoyle wrote:
Matthew Asham wrote:
; north america enum
exten = _1NX,1,Playback(doing-enum-lookup)
exten = _1NX,2,EnumLookup(${EXTEN})
exten = _1NX,3,BackGround(enum-lookup-successful)
exten = _1NX,4,Dial(${ENUM},30,tr)
exten = _1NX,5,Hangup
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup
Interesting.. how does it know to go to '6', or does it just jump +4
on failure?
That reminds me I seriously need to restructure my extensions.conf...
there's
no way currently I could add anything like that without major surgery
(only
discovered the 'local' target this afternoon so I have everything
copied/pasted).
Tony
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Re: [Asterisk-Users] RxFAX generates no tiff file
Hi Troy, People had a lot of problems like this with earlier versions of spandsp. However, the latest version is pretty solid, and people are using it in high volume production applications. If you are getting these bad results with the latest version I would be interested to see the audio log file, so I can investigate the reason. Regards, Steve Troy Settle wrote: Dunno about not being able to generate a tiff, I got rxfax to do that, but they're badly malformed. http://roanoke-voip01.psknet.com/fax/ -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip call using name in sip.conf
[foo] type=friend I do not beleive that will work for type=friend. If you use separate type=peer and type=user blocks in sip.conf it may work. Expecially if you also specify a port in the Dial(). Else, use the hostname (or a const). hmmm. then, how do i let it be dynamic if it has two blocks in sip.conf, one for inbound and one for out? i.e, how does it register its ip address in both? randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] e164.org
Hi Tony, it is the same duane - lol you are hardly allowing it to perform least cost routing, it just does one check for ip to ip call then drops back to whatever you have written on your asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Hoyle Sent: Sunday, 23 May 2004 11:34 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] e164.org Dean Collins wrote: Tony, as per you inference that e164 are up to something shady, you should talk to one of the founders Duane, he currently has about 5 open If it's the same duane who runs cacert he probably means well... however having read the site I'm still not sure whether i'd use it myself (it means trusting an external database to produce a least cost route.. I'm just not that trusting). Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems using Adtran 750 FXO and TE405P
Hello, I am trying to get an Adtran 750 w/ 1 Quad FXO and 1 Quad FXS to work with a TE405Pand I am having a few problems. I have the FXO on channels 1-4 and the FXS on channels 5-8. I have a single analog phone set connected to the first port on the FXS (channel 5) and an analog line connected to the first port of the FXO (channel 1). The FXS sees to be working fine. I can call the demo server and back and forth with SIP phones, but I cannot get anything to connect out to the CO line. I added these lines to zaptel.conf:span=1,0,0,esf,b8zsfxsks=1-4fxols=5-8unused=9-24 I added these lines to zapata.conf: context=localgroup=1signalling=fxs_kschannel=1-4 context=localgroup=2signalling=fxo_lschannel=5-8 I have also tried configuring channels 1-4 as fxsls and fxsgs, but nothing seems to work. BTW, the Adtran is brand new and according to the document the FXO ports are automatically provisioned as FXO loop start. I have attempted to connect to the admin port to verify the provisioning, but after getting no response after 8 minutes, I decided to trust the documentation. Any suggestions would be greatly appreciated. Thanks, Patrick-- This message has been scanned for viruses and dangerous content and is believed to be clean.
[Asterisk-Users] CallerID and AON in Eastern Europe
Hello All, Does anyone tried to use CallerID in Eastern Europe (Russia/Ukraine)? Our teleco provides CallerID, as well as AON, then can send _callerid_, as well as AON signals non of those 2 works on TDM400P card with FXS ports. They are using Siemence systems. How can I debug this and decode? Does anyone tried to implement AON? Thank you, Vasyl Rublyov ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] e164.org
Billy Huddleston wrote: 'local' target? What's that? http://www.voip-info.org/wiki-Asterisk+local+channels It's like a subroutine, so you can use it to call bits of the dial plan that get repeated a lot, like dialing FWD after first setting the caller ID. (AFAIK anyway... not tried to get them working yet). Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P HDLC configuration
Just would like to add, of course if it is going to help:
I am using Linux 2.4.26 on Linux, compiled from sources and latest zaptel sources.
We have T1 Internet from Verizon.
... any help appreciated.
= Original message ===
From: Vasyl Rublyov [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sat, 22 May 2004 22:09:28 -0400
Subject: [Asterisk-Users] T100P HDLC configuration
Thank you, Michael
I tried to switch to FR mode... but it did not help. I tied DLCI as 16
and 99... the same result.
I attached one more full config from Netopia and from my Linux+Zaptel
T100P systems.
DEVICE=hdlc0
# MODE=hdlc
# MODE=cisco
MODE=fr
NETMASK=255.255.255.252
GATEWAY=REMOTE_IPADDR
# FR
FR_LMI=ansi
FR_PVC=pvc0
FR_DLC=16
# FR_DLC=99
case $1 in
'start')
echo Loading T1/HDLC modules...
/sbin/modprobe zaptel
/sbin/modprobe wct1xxp
/sbin/modprobe hdlc
/sbin/modprobe syncppp
/sbin/ztcfg -vvv
echo -n Configuring HDLC interfaces, with mode
\${MODE}\
if [ ${MODE} == hdlc -o ${MODE} == cisco ]; then
echo ...
/sbin/sethdlc ${DEVICE} ${MODE}
/sbin/ifconfig ${DEVICE} ${LOCAL_IPADDR}
pointopoint ${REMOTE_IPADDR}
/sbin/route add -net ${NETWORK} netmask
${NETMASK} ${DEVICE}
echo Configuring default gateway...
/sbin/route add default gw ${GATEWAY} metric 1
${DEVICE}
elif [ ${MODE} == fr ]; then
echo , LMI \${FR_LMI}\...
/sbin/sethdlc ${DEVICE} ${MODE} lmi ${FR_LMI}
/sbin/sethdlc ${DEVICE} create ${FR_DLC}
/sbin/ifconfig ${DEVICE} up
echo Configuring Frame-Relay PVC
\${FR_PVC}\...
/sbin/ifconfig ${FR_PVC} ${LOCAL_IPADDR}
pointopoint ${REMOTE_IPADDR}
/sbin/route add -net ${NETWORK} netmask
${NETMASK} ${FR_PVC}
echo Configuring default gateway...
/sbin/route add default gw ${GATEWAY} metric 1
${FR_PVC}
else
echo , unknown mode...
fi
;;
'stop')
echo Unloading default gateway...
/sbin/route del default
echo -n Unloading HDLC configuration.
if [ ${MODE} == hdlc -o ${MODE} == cisco ]; then
echo , hdlc/cisco mode...
/sbin/route del -net ${NETWORK} netmask
${NETMASK} ${DEVICE}
/sbin/ifconfig ${DEVICE} down
elif [ ${MODE} == fr ]; then
echo , frame-relay mode...
/sbin/route del -net ${NETWORK} netmask
${NETMASK} ${FR_PVC}
/sbin/ifconfig ${FR_PVC} down
/sbin/sethdlc ${DEVICE} delete
/sbin/ifconfig ${DEVICE} down
else
echo , unknown mode...
fi
echo Unloading T1/HDLI modules...
rmmod wct1xxp zaptel hdlc syncppp
;;
'restart')
$0 stop
sleep 1
$0 start
;;
*)
echo usage $0 start|stop|restart
esac
# $ show config
# frame-relay lmi type ansi
# frame-relay tim none
# hardware acceleration enable yes
# ip gateway REMOTE_IPADDR
# ip route 0.0.0.0/0 REMOTE_IPADDR low
# interface t1 1 buildout 0-0.6
# interface t1 1 channels count 24 start 1 contiguous rate 64k
# interface t1 1 clock source network
# interface t1 1 diagnostic mode normal
# interface t1 1 dle hdlc
# interface t1 1 encoding b8zs
# interface t1 1 framing esf
# interface t1 1 operation mode hdlc
# interface t1 1 priority-queuing enable yes
# interface t1 1 pvc 1 yes
# interface t1 1 pvc 1 enable yes
# interface t1 1 pvc 1 tag Circuit 1
# interface t1 1 pvc 1 vpi 0
# interface t1 1 pvc 1 vci 35
# interface t1 1 pvc 1 cp default
# interface t1 1 pvc 1 voice no
# interface t1 1 pvc 1 pcr 0
# interface t1 1 prm-enable no
# interface t1 1 rfc1973 enable no
# interface t1 1 rfc1973 dlci 16
# interface t1 1 rfc1973 lmi none
# interface t1 1 cell-format scrambled
# interface t1 1 unused cell-format idle
# interface t1 1 ds0-autodetect no
# cp 1 yes
# cp 1 tag 36.HCGA.101976.VA
# cp 1 enable yes
# cp 1 dle hdlc
# cp 1 ip enable yes
# cp 1 ip address local LOCAL_IPADDR/30
# cp 1 ip address remote REMOTE_IPADDR/30
# cp 1 ip addressing numbered
# cp 1 ip dhcp client mode standard
# cp 1 ip mask local 255.255.255.252
# cp 1 ip mask remote 255.255.255.252
# cp 1 ip nat enable no
# cp 1 ip nat map-list Easy-PAT List
# cp 1 ip nat server-list Easy-Servers
# cp 1 ip negotiate-lan no
# cp 1 ip netbios proxy enable no
# cp 1 ip rip receive both
# cp 1 ip rip transmit no
# cp 1 ip multicast-fwd yes
# cp 1 interface-group primary
# ;Netopia 4622
# name
# preferences changes immediate yes
# preferences console default menu
# preferences date format mm/dd/yy
# preferences output format verbose
# preferences output mask bits
# preferences time format 24-hour
#
#
=
--
Thanks and regards,
Vasyl Rublyov
=End of Original message =
Re: [Asterisk-Users] e164.org
Dean Collins wrote: Hi Tony, it is the same duane - lol you are hardly allowing it to perform least cost routing, it just does one check for ip to ip call then drops back to whatever you have written on your asterisk. So eg. if I've registered 3 different sip providers and an IAX provider, plus a couple of landlines what is it doing? I guess I'm missing the point somewhere. The way I understand it is you pass it a phone number and it gives you a prefferred route to that number, which may be VOIP and may be POTS or from the looks of it MSN and lots of other things (including ldap???!!). You then pass that result straight into a Dial command, which means it could potentially do absolutely anything, including call the chinese speaking clock at peak rate. TBH I'd prefer a web page where you typed the number and it listed the alternatives (in perference order if it liked) so I could make the decision myself. Using DNS for this seems to be overkill. Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFAX generates no tiff file
Hi Mike, How do you run rxfax? You problem is probably something to do with that. Your's is the first report I have had of no TIFF file whatsoever. Regards, Steve Mike Heininger wrote: Am 22.05.2004 um 20:09 schrieb Troy Settle: Dunno about not being able to generate a tiff, I got rxfax to do that, but they're badly malformed. This is more than I get ;-) Does the fax on the other side get a success message? I get fax-rx-audio and fax-tx-audio files in /tmp but no tiff output file. Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip call using name in sip.conf
At 7:31 PM -0700 on 5/22/04, Randy Bush wrote:
[foo]
type=friend
I do not beleive that will work for type=friend. If you use separate
type=peer and type=user blocks in sip.conf it may work. Expecially
if you also specify a port in the Dial().
Else, use the hostname (or a const).
hmmm. then, how do i let it be dynamic if it has two
blocks in sip.conf, one for inbound and one for out?
i.e, how does it register its ip address in both?
randy
Short answer: you can't.
Long answer: there might be other ways around this, but I haven't
really sat down and tried to do it the right way.
Longer answer: Yes, you can, but bo is it ugly.
It appears that you have a situation where you have two Asterisk
servers. One * (we'll call it #1) is on a static IP address, while
the other (#2) moves around and is dynamically allocated by DHCP or
some other method.
You have a group of numbers that you'd like to always route from #2
to #1 when dialed. This isn't a problem, since #1 has a static IP
address, and you can just reference it with the host=1.2.3.4 entry
in your peer statement in sip.conf.
Now, going the other way around is more difficult. #1 doesn't know
the IP address of #2. There is the concept of register= in
sip.conf, but that only registers _individual user-agents_ and does
not allow one server to know that another server is at a particular
IP address. REGISTER typically is not used for server-to-server
notification of layer 3 presence (though maybe it is - it's possible
that is supported in the RFC, but I'm too lazy right now to go
digging. It's not supported in Asterisk, so that's the point here.)
So, you're out of luck I think.
There is one way to do this the way you want, and to even talk about
it makes my hair stand on end. You _could_ put the real called
number into the caller ID name (SetCIDName) and then send it to a
single registered extension. That registered extension would take the
call in on the other side, parse out the $CALLERIDNAME value and then
set an ${EXTEN} based on the results and proceed with the dial path.
Errgh - I need to go take a shower now.
NOTE: You'd of course have to route all of your RTP through #1 in
order for it to get to #2, and re-invites and all that neat stuff
will fail, because you're tunnelling SIP over SIP.
Last note: you might be able to write a hack that pulled this data
out of the SIP database, kind of like a very strange adaptation of
default-network in IOS - beacon routes. Here's another one of my
hypothetical program descriptions for this mythical application:
app1*CLI show application GetHostIP
-= Info about application 'GetHostIP' =-
[Synopsis]:
Terrible kludge to get IP address of remote Asterisk servers.
[Description]:
GetHostIP(proto/username): Looks up the given name in the protocol
username list and returns the IP address of where that username is
currently registered in variable ${DYNAMICADDR}. Useful for setting
up beacon accounts that register from dynamically-addressed
Asterisk hosts so one can trunk calls to them. Currently supports
IAX/IAX2/SIP protocols.
app1*CLI
I just come up with the ideas, I don't program 'em.
JT
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