[asterisk-users] Asterisk crashes
Hello, I have very annoying problem with asterisk 1.4.4: Every evening when I have peak load asterisk crashes, peak load is only over 20-30 sip-to-h323 simultaneous calls. Nothing special in logs after crash. Load average never was higher than 0.3, asterisk never uses more than 12% CPU (according to top). Tried SVN versions - same result. Both h323 and sip peers has only one codec allowed - g729 - so no conversion. There is no conferences, call recordings or something like this - very simple setup. Software config: Linux Slackware 11.0 Kernel 2.6.21.1 Asterisk 1.4.4 (native h323 channel from asterisk tarball) Libpri-1.4.0 Zaptel-1.4.2.1 (using ztdummy for internal sync, no zaptel hardware) pwlib_v1_10_0 openh323_v1_18_0 Hardware config: Intel SE7210TP1 motherboard P4 3GHz HT 1Mb cache CPU 1Gb RAM (dual channel, two same DIMMs from intel recommended list) 80Gb SATA HDD No zaptel hardware or even any PCI cards There isn't overheating and voltage problems with a hardware (controlling over IPMI), this hardware (with another HDD and software versions) worked fine about year with asterisk restarts manually only for a version upgrade. Could somebody point me the way to debug this problem? Thank you! -- Sincerely, Elman Efendiyev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP-H323 calls without proxying RTP
Hello, Could somebody tell me is it possible to use asterisk without RTP proxying in SIP-H323 calls? I mean exactly what canreinvite=yes option do in SIP-SIP calls. I don't need a transcoding, only a signaling conversion, and this is possible with some softswitches, so i wondering what about asterisk. Same question about H323-H323 calls I'm using NuFone Network's H323 cahhel Thanks -- Sincerely, Elman Efendiyev PROTECH INC. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Digium TE110P
Hi, Thanks for info but could You please tell an exact mobel name of your motherboard? About led - when PRI cables not connected to TE100P or when there is a problem no physical level whth PRI led should be red blinking. When PRI link connecteg successfully led should provide a green continuous light -- Sincerely, Elman Efendiyev PROTECH INC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of rivoli.durand Sent: Friday, 16 February, 2007 15:52 To: asterisk-users Subject: [asterisk-users] Digium TE110P Hi I am currently installing a TE110P. SUSE10 The zttest test result is : average 99.9991%. My server : processor Intel® Celeron® D 330, 2.66 GHz, cache 256 Ko, FSB 533 MHz , 1G RAM. Hope it can help. Now I have a question to TE110P users : The card is physically plugged, modprobe, ztcfg ok etc ... There is a red led blinking. The question is : should this led provide a green continuous light or is it correct to have a red blinking light ? Thanks in advance Olivier Envoyez vos cartes de voeux depuis www.laposte.net Elles seront ensuite distribuées par le facteur : pratique et malin ! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Trixbox vs. Custom install
Hi, I'd recommend if you need quick and easy setup - use [EMAIL PROTECTED] or Trixbox or something like this, and if you need customized setup and want to understand system in detail - use your favorite distribution and setup * from sources. I'm prefer Slackware for any * installation, but your coise on your own. -- Sincerely, Elman Efendiyev PROTECH INC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefano Corsi Sent: Monday, 12 February, 2007 18:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Trixbox vs. Custom install Hello, I'm following the thread [EMAIL PROTECTED] vs Trixbox, and I have a similar question: if someone is going to install Asterisk, FreePBX and A2Billing, should you advice him/her to use Trixbox ... or a custom step by step installation on a distribution of his/her choice? Thanks Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE110P working hardware configurations
Helo, I have a troubles getting to stable work of Digium TE110P card (mailed some time earlier in the list) - I can't get 100% pseudo zap interface accuracy (zttest), so getting HDLC aborts and call drops. I tried number motherboards, hardware and software configs according to info in wiki, thisl list and number of websites - no luck. So I ask everyboby who successfully use Digium TE110P card and get stable 100% pseudo zap interface accuracy with zttest to put exact hardware (motherboard model, NIC, CPU, RAM, HDD) and software (kernel and zaptel versions, maybe zaptel.conf and zapata.conf) configuration here. I think this info could be very useful for many people and we can make detail compatibility list in wiki. There is some hw examples list in wiki already but I think not enough detailed. Thank you! -- Sincerely, Elman Efendiyev PROTECH INC. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium TE110P
at e800 Memory at fe6fd000 (32-bit, non-prefetchable) Capabilities: [40] Power Management version 2 I appreciate any ideas. Thank you. -- Sincerely, Elman Efendiyev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...)
Are you trying to send fax over T.38? As far I understand * don't support T.38 event when passing packets trouth. I'm interested in T.38 support too, so if anybody could explain why * can't just pass theese packets (as i undrstand there is no need foe recoding etc.) I would be very appreciative. Are anybody currently working on T.38 support for * ? I don't mean T.38 support on zap interfaces, just passing T.38 packets trouth asterisk -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai Militzer Sent: Tuesday, November 23, 2004 6:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...) Hello everyone! I tried to send a fax over SIP with an Asterisk Server in the middle (no Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway is external). Whenever I start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following messages start to appear (about 100 of them) Nov 23 16:27:25 NOTICE[1061908]: rtp.c:489 ast_rtp_read: Unknown RTP codec 127 received After that asterisk gets totaly confused: Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:35 NOTICE[1061908]: rtp.c:489 ast_rtp_read: Unknown RTP codec 1 received Nov 23 16:27:35 WARNING[1061908]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (15)? Nov 23 16:27:35 NOTICE[1061908]: channel.c:1724 ast_set_read_format: Unable to find a path from G723 to ALAW Nov 23 16:27:35 NOTICE[1061908]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G723 Nov 23 16:27:35 WARNING[1061908]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (20)? Nov 23 16:27:35 WARNING[1061908]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/sip.westend.com-082fd1b8(8) to SIP/xxx-3ef8(1) Nov 23 16:27:35 WARNING[1061908]: channel.c:2633 ast_channel_bridge: Can't make SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 compatible Nov 23 16:27:35 WARNING[1061908]: res_features.c:358 ast_bridge_call: Bridge failed on channels SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 == Spawn extension (macro-enumcall, s, 211) exited non-zero on 'SIP/sip.westend.com-082fd1b8' in macro 'enumcall' == Spawn extension (xxx, 911879, 7) exited non-zero on 'SIP/sip.westend.com-082fd1b8' -- Executing NoOp(SIP/sip.westend.com-082fd1b8, ) in new stack cdr_odbc: Query Successful! Then the call gets hung up. I cannot explain why this happens. I would have explanations for the fax-machines not able to synchronize or faxes not being transmitted correctly, as the communication is SIP only, but this seems a bit strange to me. I can't even tell, if my asterisk produces these messages, or the other side (aka PSTN-Gateway). If anyone can bring some light into this behavior I would be very greatful. Best regards Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Ltticher Strae 10 Tel 0241/701333-11 [EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879 ___ 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] Fax over SIP Problems (sorry for this topic ...)
Log you posted looks like sitll T.38 problemm Which gates you use? Gateways able to support T.38 will try to use it by default no matter what codec in use. I'd suggest check gateway setup if T.38 is completely disabled Fax call with G711 passtrouth (without T.38) havent any difference comparing to voice call, you can (and probaby will) have troubles with fax transmission (quality, line drops etc) but not with * complain about codecs. -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai Militzer Sent: Tuesday, November 23, 2004 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...) Hello! Elman Efendiyev wrote: Are you trying to send fax over T.38? As far I understand * don't support T.38 event when passing packets trouth. No, I'm not trying to send the fax over T.38, I am trying to send it in the voice path by using the G711 alaw codec. This should work, I think, but it doesn't. Best regards Kai ___ 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] H.323 call problemm (no sound)
Hi all,
I'm having trouble with H.323 outbound calls, * connects but there is no
sound in both ways.
I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which
using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729
licenses installed and this is onli one call.
I tested my * with another ITSP over SIP and G.729 codec and there was
all ok
Here is my configs and consile output:
extensions.conf:
[test]
exten = _,1,Dial,H323/[EMAIL PROTECTED]
h323.conf:
[general]
port = 1720
bindaddr = 0.0.0.0
tos=lowdelay
disallow=all
allow=g729
context = test
console output:
*CLI h.323 debug
H323 debug enabled
*CLI -- Executing Dial(SIP/234-d01b,
H323/[EMAIL PROTECTED]) in new stack
Allowed Codecs:
Table:
G.729A{sw} 1
G.729{sw} 2
Set:
0:
0:
G.729A{sw} 1
G.729{sw} 2
-- Making call to [EMAIL PROTECTED]
== New H.323 Connection created.
-- 234 is calling host [EMAIL PROTECTED]
-- Call token is ip$localhost/17498
-- Call reference is 17498
-- Called [EMAIL PROTECTED]
-- Sending SETUP message
-- Ringing phone for 111.222.111.222
-- H323/111.222.111.222 is ringing
=*= In CreateRealTimeLogicalChannel for call 17498
-- externalIpAddress: 111.111.111.111
-- externalPort: 19822
-- SessionID: 1
-- Direction: IsReceiver
-- Started logical channel: receiving G.729{sw}
-- channelsOpen = 1
=*= In CreateRealTimeLogicalChannel for call 17498
-- externalIpAddress: 111.111.111.111
-- externalPort: 19822
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.729A{sw}
-- channelsOpen = 2
-- Connection Established with Unknown [111.222.111.222]
-- H323/111.222.111.222 answered SIP/234-d01b
channelsOpen = 1
*CLI -- Received RELEASE COMPLETE message...
-- Sending RELEASE COMPLETE
channelsOpen = 0
-- Unknown [111.222.111.222] has cleared the call
== Spawn extension (test, , 1) exited non-zero on
'SIP/234-d01b'
== H.323 Connection deleted.
Could anybody point me what I'm doing wrong
Thanks.
--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED]
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RE: [Asterisk-Users] H.323 call problemm (no sound)
Could You please tell me which exactly version of H.323 (or source files date or so) I need for latest cvs I tried last versions before march (cvs checkout -D 2004-03-01) and before april (cvs checkout -D 2004-04-01), replace files in channels/h323 directory of last CVS with files from theese versions. H323 compiles but asterisk gives an error: chan_h323.c: In function `load_module': chan_h323.c:1975: warning: passing arg 7 of `h323_callback_register' from incompatible pointer type chan_h323.c:1975: error: too many arguments to function `h323_callback_register' make[1]: *** [chan_h323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Monday, September 20, 2004 5:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] H.323 call problemm (no sound) [EMAIL PROTECTED] wrote: Hi all, I'm having trouble with H.323 outbound calls, * connects but there is no sound in both ways. I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729 licenses installed and this is onli one call. I tested my * with another ITSP over SIP and G.729 codec and there was all ok Here is my configs and consile output: we use older version of h323 driver on latest CVS.. this way it works fine. i think you need to look for h323 version before may/april '04 Jeremy, would that be correct? SJ ___ 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 would you handle a fax without T.38 orG.711uLaw?
Isn't it possible to use T.38 for interconnecting hardware gates supporting T.38 with asterisk using SIP REINVITE? I'm not shure but but think its's might be possible because after reinvite traffic goes directly from one gate to anotger, not over Asterisk -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Friday, September 17, 2004 1:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How would you handle a fax without T.38 orG.711uLaw? I'm not sure you can. Isn't the problem to do with the slicing of the data. Ulaw does it such that a fax can survive but others don't? If someone else knows better then I'd love to know how. I want to change my upstream codec to GalaxyVoice which is currently ULAW for something skinnier like G729. Currently my need to send faxes forces my down the G711 road. Mark [EMAIL PROTECTED] said: Let's say you were wanted to terminate calls onto your Asterisk system but your only available codec was G.729 and you had no control over the remote SIP proxy sending you the traffic. What would you do? Does anyone have an update on Asterisk supporting T.38 with SIP? Thanks! chris___ 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 -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ 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 would you handle a fax withoutT.38orG.711uLaw?
And what about using same codecs for asterisk and endpoints? Lets say G.729. Yes, it needs license but while G.729 is industry standart de-facto I thing most of us need to use it anyway -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Sikkema Sent: Friday, September 17, 2004 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] How would you handle a fax withoutT.38orG.711uLaw? [EMAIL PROTECTED] wrote: Isn't it possible to use T.38 for interconnecting hardware gates supporting T.38 with asterisk using SIP REINVITE? I'm not shure but but think its's might be possible because after reinvite traffic goes directly from one gate to anotger, not over Asterisk We've seen a problem here with asterisk. Wehn Asterisk sends it reinvite, it uses its own codecs, not those of the other endpoint. So until someone fixes that (when possible), there's no way this will work. We're using a CVS version of approx. a month ago. -- Andreas SikkemaRits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] Unknown RTP codec 72 received
Hi all, I get Unknown RTP codec 72 received message in console when call in progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN over voicepulse connect (IAX) and to FWD echo test (SIP). But this message only with one SIP client, others (X-Lite too) not giving this message. All X-Lite settings are identical. Asterisk is last cvs version This what I see in console (FWD): vgw3*CLI -- Executing Dial(SIP/332-552e, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/fwd-357f is ringing -- SIP/fwd-357f answered SIP/332-552e -- Attempting native bridge of SIP/332-552e and SIP/fwd-357f Sep 13 11:02:52 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:02:57 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:03:01 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:03:02 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:03:07 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received == Spawn extension (full-access, 1700613, 1) exited non-zero on 'SIP/332-552e' And voicepulse: vgw3*CLI -- Executing Dial(SIP/332-3f30, IAX2/[EMAIL PROTECTED]/011) in new stack -- Called [EMAIL PROTECTED]/011 -- Call accepted by 66.234.228.160 (format GSM) -- Format for call is GSM -- IAX2/voicepulse/3 stopped sounds -- IAX2/voicepulse/3 stopped sounds -- IAX2/voicepulse/3 answered SIP/332-3f30 Sep 13 11:06:37 NOTICE[262160]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:06:42 NOTICE[262160]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received same skipped-- Sep 13 11:10:24 NOTICE[262160]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:10:29 NOTICE[262160]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received -- Hungup 'IAX2/voicepulse/3' == Spawn extension (full-access, 492103998463, 1) exited non-zero on 'SIP/332-3f30' My sip.conf (234 user didn't give Unknown RTP codec 72 received message, 332 gives this message, only difference is internet path to users and firewall type): [general] port = 5060 tos=lowdelay videosupport=no disallow = all allow = gsm allow = ulaw canreinvite = no [fwd] context = in type = peer disallow = gsm allow = ulaw userneme = XX secret = xxx host = fwd.pulver.com [234] context = full-access type = friend disallow = ulaw insecure = no username = 234 secret = xxx host = dynamic nat = yes dtmfmode = rfc2833 callerid = 234 [332] context = full-access type = friend disallow = ulaw insecure = no username = 332 secret = xxx host = dynamic nat = yes dtmfmode = rfc2833 callerid = 332 Could somebody tell me whay this Unknown RTP codec 72 received means and how to fix it? Thanks. -- Sincerely, Elman Efendiyev [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] Unknown RTP codec 72 received
Thanks for the hint Eric, but yes, before sending a message to the list I checked google and wiki and NO - I didn't find an answer/solution/any info on this subject. There was couple of the same questions on the list but none of them answered -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Monday, September 13, 2004 5:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Unknown RTP codec 72 received On Mon, 2004-09-13 at 06:13, Elman Efendiyev wrote: I get Unknown RTP codec 72 received message in console when call in progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN None of the 19 hits I saw on Google about this were helpful? macroTo search the Asterisk mailing list archive go to www.google.com and put site:lists.digium.com in addition to your other query terms./macro -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] T.38 pass-thru
Hello, As I understand * don't supports T.38 in Zap channels (please correct me if I'm wrong, BTW is there plans for such support?) I believe it's should support T.38 in pass-thru mode. I mean setup like this: Hardware gate with T.38 -- Asterisk -- Hardware gate with T.38 But I had troubles with this setup (no faxing) while two gates conneted directly with same network path without Asterisk able to faxing without problemms Where I'm wrong? -- Sincerely, Elman Efendiyev [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] T.38 pass-thru
Jusc couldn'n transmit faxes trouth asterisk. It just hangs up when starting a fax transfer If U will do some experiments with it I would be happy to hear any reslts/info Thanks -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tracy R Reed Sent: Monday, September 06, 2004 10:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] T.38 pass-thru On Mon, Sep 06, 2004 at 10:40:44AM +0300, Elman Efendiyev spake thusly: I believe it's should support T.38 in pass-thru mode. I mean setup like this: I have seriously been considering giving this a try lately also. I will be very interested to see how it works out. What sort of problems exactly did you run into? -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig ___ 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] FXO/FXS with T.38 over SIP
Hello, Could anybody suggest cheap FXO/FXS devices with full T.38 support over SIP? I found a number of devives with declared H323/SIP and T.38 support but some of them supports T.38 only with H323, others have buggy T.38 -- Sincerely, Elman Efendiyev [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
[Asterisk-Users] Phone numbers for testing
Hi, Is there any phone numbers with answering machine wich can record my voice and play it back to me? It would be very helpful for asterisk testing, but im not shure such service exsists at all. -- Sincerely, Elman Efendiyev [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] Phone numbers for testing
Yes, something like this. I would like ability to call such system via PSTN to test my * setup and my ITSP termination Something like this: My* -- MyITSP -- PSTN -- System with extensions you tell about -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, September 03, 2004 12:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Phone numbers for testing Is there any phone numbers with answering machine wich can record my voice and play it back to me? It would be very helpful for asterisk testing, but im not shure such service exsists at all. Do you mean somthing like: ; record a temporary GSM file exten = 3920,1,Wait(1) exten = 3920,2,Record(/tmp/ast:gsm) exten = 3920,3,Wait(1) exten = 3920,4,Playback(/tmp/ast) within your extensions.conf? ___ 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] D-Link DG-104SH H323 problemm
Hi, I'm using D-Link DG-104SH (H323 4 port FXS gateway) with analig phone connected to it and X-Lite softphone as endpoints with * When I calling from X-Lite to analog phone it's ok When I dilaling X-Lite from analog phone, X-Lite si ringind but whei I picked up X-Lite connection drops IP of DG-104SH is 192.168.1.3, H323 ID is GW1 X-Lite number is 233 Here is * output: -- Executing Dial(H323/ip$192.168.1.3:2406/14522, SIP/233) in new stack -- Called 233 -- SIP/233-a652 is ringing -- SIP/233-a652 answered H323/ip$192.168.1.3:2406/14522 == Spawn extension (h323, 233, 1) exited non-zero on 'H323/ip$192.168.1.3:2406/14522' Extensions.conf -- [general] static=yes writeprotect=no [h323] exten = _233,1,Dial(SIP/233) [sip] exten = _7025818,1,Dial(h323/[EMAIL PROTECTED]:1720) --- H323.conf --- [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw gatekeeper = DISABLE [GW1] type=friend ; I tried type h323 and user here to host=192.168.1.3 context=h323 - Sip.conf [general] port = 5060 allow = ulaw allow = gsm [233] context = sip type = friend insecure = no username = 233 secret = mysecret host = dynamic dtmfmode = rfc2833 callerid = 233 When I using DG-104SH and netmeeting (without asterisk) its ok in both directions Could anybody explain what I'm doing wrong? Thanks -- Sincerely, Elman Efendiyev [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] Busydetect problems
In my case this was because PSTN line didn't reverse polarity for disconnect notification and busy tone level was to low for asterisk to detect it. I solved this problemm by increasing rxgain (up to 4 in my case, your may de different) -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fabricio Chicon Sent: Monday, July 26, 2004 5:59 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Busydetect problems Hi guys. I have a XP100P Clone , and the busydetect dont work for me.. PSTN---Asterisk---Sip---AsteriskPBX Any call from pstn side dont disconnect ... I have no disconnect supervision and busydetect dont work... Please Help me. Zapata.conf [channels] echocancel=yes usecallerid=no hidecallerid=no rxgain=0.0 txgain=0.0 signalling=fxs_ks callprogress=no context=entrada channel=1 musiconhold=default busydetect=yes busycount=7 ;echocancel=yes ;usecallerid=yes ;hidecallerid=no I try compile asterisk with BUSYDETECT_MARTIN and change dsp.c for my busy tone but no results... Sorry my bad english. Fabricio Chicon Pereira da Silva [EMAIL PROTECTED] http://www.freenetworks.com.br ___ 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] PBX functions and different channels grouping
Hi All, I need to replace old analog PBX with Asteriskl and X-Lise SIP SoftPhones as client phones. First: I have problems with implementation of PBX functions. I need and unsuccesfully tried theese functions (took info at http://voip-info.org/wiki-Asterisk+PBX+functions) Call Pickup: Supported in the standard installation (*8 - defined in res_parking.c +54) - Just don't understand how to define pickup groups Unattended Transfer (or blind transfer): Implemented in Asterisk (#), optionally also in the phone - when I press # on X-Lite it hangs up Attended transfer: Implemented in Asterisk (FLASH) - How nj make FLASH on X-Lite? Call Pickup: Supported in the standard installation - How to use it? Automatic Call Distribution: ACD - is it possible at all? Also I tried Transfer command like this: exten = 9,1,Transfer(SIP/234) And when I press 9 got thiss error: -- Executing Transfer(SIP/233-b6ad, SIP/234) in new stack Jul 24 16:34:09 WARNING[213006]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Where 192.168.1.1 IP of SIP client calling from (233 phone) Second: Is it possible to group different channels? I need thing like this: Try to call ZAP channel, if it's busy (or another problemm with it) then try to call H323 channel, if busy again try to call IAX2 channel I found info only for ZAP channel grouping and dialing channels simultaneously Please help me with theese problemms Thanks! -- Sincerely, Elman Efendiyev [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] Please help I fear I have missed something very important! but what?
Looks like you missed 's' extension for incoming calls
You need something like this in extensions.conf
exten = s,1,Answer
exten = s,2,Dial(SIP/1001,20,t)
See sample of extensions.conf in asterisk distribution (make samples if
you didn't install samples)
--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sales
Sent: Saturday, July 24, 2004 9:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Please help I fear I have missed something
very important! but what?
Sorry about this, I have been struggling with the basics of my asterisk
config.
I set up two sip peers and two phones. And I set up lots of dial masks
for outgoing calls, all my outgoing calls were working great, however
incoming calls were a different matter altogether, I cannot get incoming
calls to work. So I have gone back to a very basic FWD config, with one
phone which as far as I am aware should work, but doesn't. I cannot find
info on how to fix this.
Below is my sip.conf
[general]
port = 5060
bindaddr = xxx.xxx.xxx.xxx
context = sip
register = 2:[EMAIL PROTECTED]/1001
[fwd]
type=friend
secret=xx
username=xx
host=fwd.pulver.com
;
;
[1001]
type=friend
username=xx
host=dynamic
secret=xxx
callerid=Home 1001
dtmfmode=RFC2833
mailbox=1001
context=sip
and here is my extensions.conf:
[general]
static=yes
writeprotect=no
;
[globals]
HOME=SIP/1001
;
[sip]
exten = 1001,2,Dial(SIP/1001,20,t)
include = fwdnet
;
[fwdnet]
exten = _8.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED],t
Now as I said I can call out no probs by dialing 8 then the FWD number,
but incoming calls don't work, and as far as I can see that should ring
ext 1001 for 20 secs.
Could someone please help a complete Linux/Asterisk Newb, as apart from
this I have learnt a hell of a lot. But it's the last thing I need to
solve.
The linux box for this testing has a unfirewalled public IP address, so
there is no problems with NAT
Please please can someone help. If I have missed something important
then I aplogise, as I have been scouring the wiki and the archives to no
avail
Regards
Stuart Buchanan
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[Asterisk-Users] IP Soft Phone with FAX
Hi, I need to send and receive faxes over VoIP in realtime. I mean: user calls from VoIP network to fax machine on PSTN, but starts voice conversation with user B on that fax machine. Then users agree to send a fax (any direction), pressed start, completed fax transmission and then continue a voice conversation. This is one of generic ways to use analog fax machine. As I understand this can be done with analog fax machine connected to any FXS device and G.711 codec. My problemm is that I need a software phone to to use faxes, not hardware, and cannot find any software terminal able to send/receive fax in realtime, over already established connection (by voice). T38 is more elegant way for fax but anyway I cannot find any software phone able to work with is by way I need. I found some solutions for faxes (spandsp, Hylafax) but is I understand they need to make fax calls (or answer a call) by themselves, so didn't fit to my needs (please correct me if I'm wrong) I think the best solution would be a software phone with fax support by T38, but would be happy to find software phone able to faxing in realtime by any metod. Is such softwate exists at all? Can anybody suggest me a solution for my problemm? Thanks. -- Sincerely, Elman Efendiyev [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
