[asterisk-users] sip_write warning when executing Pickup of CAPI
I'm trying to pick up a ringing SIP phone (203) across the office with exten = *9,1,Pickup(783743) where 783743 is the local part of the number that our ISDN works on. I tried all of these first: exten = *9,1,Pickup(203) exten = *9,1,Pickup(SIP/203) exten = *9,1,Pickup([EMAIL PROTECTED]) and got a declined message back from my phone (snom 300), so I then switched to picking up the ringing ISDN line (it's BT ISDN2e on a pair of Eicon Diva BRI-2M cards) The Pickup(783743) works (the phone across the room stops ringing), but the calling party gets a nasty distorted noise back down the phone, and I get dozens of these messages: Dec 5 11:37:50 WARNING[26972]: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/64) Any ideas what I'm doing wrong? Asterisk is 1.2.11 on Debian with 2.6.8-2-686 kernel. Thanks Tom ___ --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] Pricing for DS3000P
Agreed, those are the figures we were able to get from Digium... I'm still waiting for a confirmation, but I'm being safe with a $4k estimate.. snip What's so special about Digium cards that makes them this expensive? $4000 for a PCB is extortion IMO! Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] does asterisk work with other processors
I have tried numerous versions of asterisk from asterisk at home to compiling it myself through the cvs server. I don't understand it works fine with the intel p2 box but not the faster via cyrix box. Is it the processor or something? Have a look in the makefile. It might be necessary to set the PROC= line. I have a VIA Epia board and it is necessary to set PROC=i586 for Asterisk to run because the kernel thinks it is an i686 - but it does not have the full i686 instruction set as far as I can tell. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] a simple call to my girlfriend
snip P.S. I don't want to use skype (not open standard, it still doens't work well in Linux and eats al the time of my old laptop CPU). Skype would do you the best. Clearly not. It won't work on his old laptop, will it. Like someone else suggested I recommend something like freeworlddialup. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gradwell UK DID + DTMF
Does anyone have a Gradwell UK SIP number successfully receiving DTMF working with their Asterisk? If so, please could you post the relevant bits of your config files. Thanks in advance Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK DID providers
Hi Can anyone provide me with a Manchester (0161) UK DID number, preferably IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume will be low. The critical thing is that DTMF must be correctly passed 100% of the time, unlike Sipgate, my current (free) provider, whose DTMF detection/passing is not at all reliable, making it useless for a virtual receptionist scenario. I don't mind paying for this service (free is good though...), as long as it is reasonably less than the cost/rental of another physical BT line in to our premises. Regards Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] UK DID providers
Thanks for your quick reply. Do you have experience with them? Does their DTMF work properly? Any chance of posting the juicy bits of your config files if you use them please? Cheers! Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Hamill Sent: 28 May 2005 14:43 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] UK DID providers On Saturday 28 May 2005 14:31, Tom Fanning wrote: Hi Can anyone provide me with a Manchester (0161) UK DID number, preferably IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume will be low. Yeh, Sipgate's price is good (hey you can't argue with £0 setup and £0 per month...) but the service is lukewarm- they drop off the 'net a fair amount and some calls just result in an unavailble tone to the caller (no incoming SIP activity). Try www.gradwell.com for an 0161... http://www.gradwell.com/voip/ddi-inbound.php They offer IAX2 recently as well as SIP. gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] CRM integration (was RE: CallerID)
snip Could you use javascript, or java from within the browser, which is both portable, and likely to work on ANY browser that way there is no installation as such, just visit the page, and leave a browser window open (minimised) which is 'listening' for connections ?? Sigh... Browsers don't listen. They inititiate a connection, process the requested transaction with the web server, and close the connection. The simply can't be used to listen for an arbitrary connection. Actually, I don't think that you are quite right here. The guy mentioned Java from within the browser. I believe that I am right in saying that a Java applet should very well be able to listen for tcp connections as well as udp datagrams. Try this primer: http://homepages.uel.ac.uk/2795l/pages/javaapps.htm#Class%20ServerSocket%20( TCP%20Server%20Connections) Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] CRM integration (was RE: CallerID)
snip Browsers don't listen. They inititiate a connection, process the requested transaction with the web server, and close the connection. The simply can't be used to listen for an arbitrary connection. Actually, I don't think that you are quite right here. The guy mentioned Java from within the browser. I believe that I am right in saying that a Java applet should very well be able to listen for tcp connections as well as udp D'oh! I had misread the PP's statement and assumed he meant a bareback browser window. You are, of course, quite right. A Java app could handle this, but we are still left with the issue of having to install SOMETHING, even if it is a small Java app, on the client to make this work. No installation as such, just make sure a Java virtual machine is present on the machine. Seconds to load. I would say that Java would be ideal for an application like this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Analog Lines
Grandstream Handytone 486 or similar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cook Sent: 24 May 2005 21:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Analog Lines I am looking for a cost effective way to drop analog lines from our asterisk system to support modems and faxes. More than would typically be done with TDMxxB cards. I have looked at going with a T1 interface to Channel Bank, but that just seems like a very expensive way to solve this problem. ($1500 - $2000 ). Any suggestions? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] MusicOnHold Loudness/Distortion
snip For whatever reason, the music on hold is extremely distorted and loud. It didn't used to be this way and I haven't changed anything, yet it persists. This is on all the channels we use (SIP, IAX2, Zap, ALSA). Can anyone help with this, or has anyone seen this? The mp3s play fine on any computer and haven't changed since they did work. Those wishing to hear for themselves, feel free to call extension 8800 at the number/addresses below. Bryce Your DTMF recognition seems screwed up. I can't get ex 8800, but I can get the MOH by dialling 80. Found that out by accidently misdialling the wrong extension. There's slight echo on your line too, and the voices sound muddy somehow. Can't help you with the dodgy MOH, sorry. Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Boosting Internet Bandwidth for VOIP
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chawki hammoud Sent: 21 May 2005 05:32 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Boosting Internet Bandwidth for VOIP There was errors when I tried to start the script recommended by Andrew to boost bandwidth for voip snip iptables v1.2.9: Couldn't load match `p2p':/lib/iptables/libipt_p2p.so: cannot open shared object file: No such file or directory Try `iptables -h' or 'iptables --help' for more information. iptables v1.2.9: Couldn't load match `ipp2p':/lib/iptables/libipt_ipp2p.so: cannot open shared object file: No such file or directory Try `iptables -h' or 'iptables --help' for more information. iptables: No chain/target/match by that name Any suggestions? Doesn't look like iptables is installed properly. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipgate incoming DTMF
Has anyone found a solution or received anything useful or constructive from Sipgate UK regarding their problems with incoming DTMF tones being filtered out of the audio stream but not passed via SIP to Asterisk, rendering IVR systems useless? It worked for a while then completely broke Cheers Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Problems with VIA Chipset
In the asterisk makefile, you need to make sure that the variable PROC ends up getting set as i586. Easiest way to do that is to remove all the conditional stuff around the PROC=xxx statements and just put PROC=i586 in its place. Works fine then. The problem I had was that linux thought that the via was an i686 when in fact the VIA doesn't support all of the i686 instructions, its closer to an i586, or so google would have me believe. Either way, it works. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Armin Lediger Sent: 11 May 2005 22:15 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problems with VIA Chipset Hi, I am trying to install asterisk 1.0.7 on a VIA EPIA 5000 board - anyone of you already managed to do so? I got V1.0.6 running, but 1.0.7 seems not to compile. I don´t want to bother you all with output code of the errors I get when I try to compile asterisk; I am just curious if anyone of you made it! Thanks for a quick reply! Sincerely, Armin Lediger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Checking for a sound file
Causes me to wonder a couple of things. Why does ANYONE use Outhouse or Outhouse Express? There are many much more friendly Windows E-mail clients, from Mozilla on down Tight integration with Exchange 2003. Find me an alternative client that is as stable and that has such tight integration and I'll jump ship immediately. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Illegal instruction (core dumped)
Hi Grabbed the most recent stable asterisk from CVS as documented here: http://www.asterisk.org/index.php?menu=download Didn't bother with zaptel or libpri as I have no Digium hardware nor T1 or E1. Did make install asterisk; make samples. Started asterisk with asterisk -c and it crashes: . . . Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_modem.so] = (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_aopen.so = (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) == Registered channel type 'Modem' (Generic Voice Modem Channel Driver) [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [res_adsi.so] = (ADSI Resource) == Parsing '/etc/asterisk/adsi.conf': Found Illegal instruction (core dumped) Build environment is Mandrake 10.1 official. Didn't have this problem on a Mandrake 10.1 Community box running in vmware - it worked perfectly the first time. Putting noload = res_adsi.so in extensions.conf just causes it to crash elsewhere during the load. Compilation worked fine except for this lot which came out of stderr. Is this normal? In file included from editline.c:18: term.c: In function `term_move_to_line': term.c:556: warning: implicit declaration of function `tputs' term.c:556: warning: implicit declaration of function `tgoto' term.c: In function `term_set': term.c:913: warning: implicit declaration of function `tgetent' term.c:931: warning: implicit declaration of function `tgetflag' term.c:940: warning: implicit declaration of function `tgetnum' term.c:943: warning: implicit declaration of function `tgetstr' term.c:943: warning: passing arg 3 of `term_alloc' makes pointer from integer without a cast In file included from editline.c:18: term.c: In function `term_echotc': term.c:1441: warning: assignment makes pointer from integer without a cast ar: creating libtime.a frame.c: In function `ast_fr_fdread': frame.c:360: warning: assignment discards qualifiers from pointer target type chan_modem_aopen.c: In function `aopen_read': chan_modem_aopen.c:327: warning: assignment discards qualifiers from pointer target type chan_modem_bestdata.c: In function `bestdata_read': chan_modem_bestdata.c:375: warning: assignment discards qualifiers from pointer target type chan_modem_i4l.c: In function `i4l_read': chan_modem_i4l.c:446: warning: assignment discards qualifiers from pointer target type chan_iax2.c: In function `__send_command': chan_iax2.c:3574: warning: assignment discards qualifiers from pointer target type app_mp3.c: In function `mp3_exec': app_mp3.c:169: warning: assignment discards qualifiers from pointer target type app_festival.c: In function `send_waveform_to_channel': app_festival.c:213: warning: assignment discards qualifiers from pointer target type app_nbscat.c: In function `NBScat_exec': app_nbscat.c:147: warning: assignment discards qualifiers from pointer target type codec_ilbc.c: In function `lintoilbc_sample': codec_ilbc.c:95: warning: assignment discards qualifiers from pointer target type codec_ilbc.c: In function `ilbctolin_sample': codec_ilbc.c:110: warning: assignment discards qualifiers from pointer target type codec_ilbc.c: In function `ilbctolin_frameout': codec_ilbc.c:128: warning: assignment discards qualifiers from pointer target type codec_ilbc.c: In function `lintoilbc_frameout': codec_ilbc.c:189: warning: assignment discards qualifiers from pointer target type codec_gsm.c: In function `lintogsm_sample': codec_gsm.c:85: warning: assignment discards qualifiers from pointer target type codec_gsm.c: In function `gsmtolin_sample': codec_gsm.c:100: warning: assignment discards qualifiers from pointer target type codec_gsm.c: In function `gsmtolin_frameout': codec_gsm.c:118: warning: assignment discards qualifiers from pointer target type codec_gsm.c: In function `lintogsm_frameout': codec_gsm.c:203: warning: assignment discards qualifiers from pointer target type src/decode.c: In function `Postprocessing': src/decode.c:25: warning: unused variable `ltmp' src/long_term.c: In function `Long_term_analysis_filtering': src/long_term.c:855: warning: unused variable `ltmp' src/long_term.c: In function `Gsm_Long_Term_Synthesis_Filtering': src/long_term.c:924: warning: unused variable `ltmp' src/lpc.c: In function `Reflection_coefficients': src/lpc.c:214: warning: unused variable `ltmp' src/lpc.c: In function `Quantization_and_coding': src/lpc.c:322: warning: unused variable `ltmp' src/preprocess.c: In function `Gsm_Preprocess': src/preprocess.c:89: warning: unused variable `lsp' src/preprocess.c:49: warning: unused variable `ltmp' src/preprocess.c:50: warning: unused variable `utmp' src/rpe.c: In function `APCM_inverse_quantization': src/rpe.c:365: warning: unused variable `ltmp'
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 9, Issue 152
On April 17, 2005 05:55 am, Tom Fanning wrote: Illegal instruction (core dumped) Sounds like you have compiled asterisk for a processor that is greater than the processor you're running on. I.e. compiled and told it to use P4 instructions when you're on a P3, or maybe even told it to use MMX on a Via processor... Have just shoved PROC=i586 in the Makefile along with some commenting to see what happens. It's compiling right now. It is indeed on a Via Epia board. Cheers Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Illegal instruction (core dumped)
On April 17, 2005 05:55 am, Tom Fanning wrote: Illegal instruction (core dumped) Sounds like you have compiled asterisk for a processor that is greater than the processor you're running on. I.e. compiled and told it to use P4 instructions when you're on a P3, or maybe even told it to use MMX on a Via processor... Have just shoved PROC=i586 in the Makefile along with some commenting to see what happens. It's compiling right now. It is indeed on a Via Epia board. Cheers Tom Worked like a charm. Posting this here for future reference. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
