[Asterisk-Users] Passthrough and reInvite
It is not clear how exactly g729 pass-through can be enabled. I have a SIP call off a gateway come into an Asterisk menu, and then I send the SIP call to another SIP gateway using Dial(). Even though codec preferences have g729 listed first, it never gets used. Both gateways have separate peer entries in sip.conf, and both have canreinvite=yes set. Can Asterisk change the media type during a re-Invite? The call is answered as g711u initially, and then Asterisk plays a menu, and then does a Dial(). I can see Asterisk doing the reInvite, but the protocol stays at g711u. 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] ATA's
On Wed, 16 Feb 2005 07:08:08 +0800, Leo Ann Boon [EMAIL PROTECTED] wrote: See my comments in line From my experience, the ATA is a very solid, dependable piece of hardware. I was told by a source in the company that OEMs for Cisco, the units are expensive because of the high quality parts being used. The web config looks crappy but otherwise where else do you find a $100 device that does SCCP/MGCP/SIP/H323? None of the competitors even come close to that level of protocol support. For developers who have to work on various protocols, the ATA is really cool. I agree. I'm using the ATA 186 and think it's great. The latest firmware also changes the web interface - it's similar to the 7905G/7912G phones now. I've also tried the Sipura SPA-2000, but had some problems with it, so the Cisco ATA is my ATA of choice now. -Shaun ___ 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] Call asterisk from perl
You can also use the manager. Take a look at http://www.voip-info.org/wiki-Asterisk+manager+API - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 6:12 AM Subject: Re: [Asterisk-Users] Call asterisk from perl Ousmane Doukara wrote: Is it possible to call asterisk from a script ? I have a script scheduled in cron and I want to be able to Dial a number from that script whenever an event occur. Look on the wiki for outgoing spool files. ___ 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] ATA's
Shaun Ewing wrote: I'm using the ATA 186 and think it's great. The latest firmware also changes the web interface - it's similar to the 7905G/7912G phones now. Did you have much trouble getting the ATA 186 working? I have one running Version: v3.1.0 atasip (Build 040211A) I have it setup and it does poll the * server but does not work to use and errors in sip. Followed the instructions on the wiki page for them and it still wants to be a pain :( Other problem is that it is in Denmark and I am in AUS :) so timming is an issue. Any advice would be appreciated. David Uzzell This is the sip debug from * end. Sip read: REGISTER sip:203.29.98.221 SIP/2.0 Via: SIP/2.0/UDP 62.79.110.156:5060 From: Test901 sip:[EMAIL PROTECTED];user=phone;tag=4212316678 To: Test901 sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Contact: Test901 sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;expires=120 User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A) Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 62.79.110.156 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 62.79.110.156:5060;received=62.79.110.156;rport=5060 From: Test901 sip:[EMAIL PROTECTED];user=phone;tag=4212316678 To: Test901 sip:[EMAIL PROTECTED];user=phone;tag=as560861ed Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 62.79.110.156:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 62.79.110.156:5060;received=62.79.110.156;rport=5060 From: Test901 sip:[EMAIL PROTECTED];user=phone;tag=4212316678 To: Test901 sip:[EMAIL PROTECTED];user=phone;tag=as560861ed Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=pbx.unifiedau.net, nonce=4a523e7e Content-Length: 0 to 62.79.110.156:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Destroying call '[EMAIL PROTECTED]' ___ 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] SIP jitter?
Adding in experimental patches willy-nilly, especially ones that have the potential to cause huge problems, confounds attempts to isolate bugs and test functionality. Mark does a pretty good job of keeping the HEAD version solid enough to use in production, as most of us running it on a daily basis can attest. What stops you from applying the patches to your own copy, and then playing with it to your heart's content--like the rest of us? It would work just like it had really been put into CVS-HEAD. less testers less bug reports for production use is stable version (asterisk doesnt have good roadmap and versioning :( ) My point exactly If we're to see this in 1.2, we need it in CVS ASAP roy ___ 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] Extra sounds (Weather)
Liaan vd Merwe wrote on Tuesday, 15 February 2005 1:37 PM: http://edgett.bc.ca/simonsays/archives/000228.html Thank you, but this is not the script. Sincerely, Trevor Hammonds ___ 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] Extra sounds (Weather)
This is the example script (extracted from that link) you will need to find a weather page for your region an then change the urls and grep statements chow L [weather.sh] #!/bin/sh WEATHER=`lynx -nolist -dump http://weatheroffice.ec.gc.ca/forecast/textforecast_e.html?Bulletin=fpcn11.cwvr | grep -A3 ^Greater Vancouver.$ | sed 's/\.\./\. /;s/km\/h/km per hour/'` echo EXEC Festival \Online Weather forecast for\ echo EXEC Festival \$WEATHER\ echo EXEC Festival \Thank you\ - Original Message - From: Trevor G. Hammonds [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 12:23 PM Subject: RE: [Asterisk-Users] Extra sounds (Weather) Liaan vd Merwe wrote on Tuesday, 15 February 2005 1:37 PM: http://edgett.bc.ca/simonsays/archives/000228.html Thank you, but this is not the script. Sincerely, Trevor Hammonds ___ 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 __ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250 ___ 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] Which IP phone to use in Australia
On Wed, 2005-02-16 at 09:23 +0800, Stuart Elvish wrote: What sort of setup is involved for the Cisco as far as config files etc? I am used to plug and play phones (Zyxel, Grandstream, HOP etc) which require minimal configuration and have no licensing issues with them. I know for the Polycom you need to get a TFTP server for XML config files running, and I believe you need something similar for Cisco phones. Actually, you can get a polycom working without a TFTP server, mine came with SIP pre-loaded, and I configured it through the web interface. It worked fine. If you want to configure more settings (fine-tuning) or deploy more than 2 or 3, then you really want server based config files. I would also strongly suggest using an FTP server rather than TFTP, security and flexibility are both much better. BTW, Polycom *don't* say you can't use their phone in/with any particular manner/software. All they say is that if the phone breaks, and it is caused by asterisk, then they won't really help you out. However, the fact that it works, and works well pretty much says it all. In my experience (I've had very limited experience with the cisco phones) the polycom phone is better than the cisco. They are equal in most ways, except the cisco phones require you to pay some silly licensing fee, and if you buy the phone second hand, then you can't use any firmware version without purchasing it extra At least polycom provide the software to download. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Strict Routing vs Loose Routing
Hello, I was interconnecting Asterisk (v1.0) with a strict router (ie, no ;lr in routes) and I think I found a bug in the way Asterisk prepare new requests inside a dialog. I'm sending some captures (ngrep) along with my comments. This is a 200 OK (INVITE) received by Asterisk = U 2005/02/10 16:41:55.065538 143.173.202.82:5060 - 143.173.202.83:5070 SIP/2.0 200 OK..Via: SIP/2.0/UDP 143.173.202.83:5070;branch=z9hG4bK42c78895..Record-Route: sip:[EMAIL PROTECTED] 202.81;ftag=as182aa61c;lr..Record-Route: sip:[EMAIL PROTECTED]:5060..From: Call Center 1 sip:[EMAIL PROTECTED] p.trdc.telenova.com.br;tag=as182aa61c..To: sip:[EMAIL PROTECTED];tag=281B1720-1D3C..Call-ID: 3 [EMAIL PROTECTED]: 103 INVITE..Contact: sip:[EMAIL PROTECTED] 37:5060..date: Thu, 10 Feb 2005 20:41:43 GMT..server: Cisco-SIPGateway/IOS-12.x..allow-events: telephone-event..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO..Content-Type: application/sdp..Conten t-Length: 257v=0..o=CiscoSystemsSIP-GW-UserAgent 2591 2076 IN IP4 143.173.194.37..s=SIP Call..c=IN IP4 143.173.19 4.37..t=0 0..m=audio 19458 RTP/AVP 18 100..c=IN IP4 143.173.194.37..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpma p:100 X-NSE/8000..a=fmtp:100 192-194.. = At this point Asterisk builds a route list for subsequent requests (record routes + contacts): sip:[EMAIL PROTECTED]:5060 sip:[EMAIL PROTECTED];ftag=as182aa61c;lr sip:[EMAIL PROTECTED]:5060 But when Asterisk sends a BYE for this call: === U 2005/02/10 16:41:57.400529 143.173.202.83:5070 - 143.173.202.82:5060 BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0..Via: SIP/2.0/UDP 143.173.202.83:5070;branch=z9hG4bK67ce783e..Rout e: sip:[EMAIL PROTECTED];ftag=as182aa61c;lr,sip:[EMAIL PROTECTED]:5060..From: Call Cente r 1 sip:[EMAIL PROTECTED];tag=as182aa61c..To: sip:[EMAIL PROTECTED];tag=281B172 0-1D3C..Contact: sip:[EMAIL PROTECTED]:5070..Call-ID: [EMAIL PROTECTED] Seq: 104 BYE..User-Agent: Asterisk PBX..Proxy-Authorization: Digest username=55512, realm=sip.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED]:5060, nonce=420bc833181bca085f953885ace7fc72c107c8e0, respo nse=75fcd1bf185ae29becd0b4f715f5cb17, opaque=..Content-Length: 0 === As we can see it is using the contact as the URI of this request, which also appears as the last route header. As per RFC3261 (16.12.1) if the next hop is a strict router (no ;lr in route header) it should use that information as the R-URI, which is not the behaviour of Asterisk. Moreover, it should include all routes learned in the request Route header (including the one that will receive the request) when we are treating with loose routers. I have changed the source code and in my test bed (at least) is working fine. Am I missing something? Regards, Chuck. __ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250 ___ 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] Asterisk exist with error
Hello ! First time I have instaled Asterisk without problem and working with a SIP clinet (X-Lite). Then I try to make the H323 with came with Asterisk. So, I DL pwlib v.1.5.2 in /root (./configure ; make) no errors DL openh323 in /root (./configure ; make opt): /root/openh323/src/h248.cxx:6178: internal error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://bugzilla.redhat.com/bugzilla/ for instructions. make[1]: *** [/root/openh323/lib/obj_linux_x86_r/h248.o] Error 1 make[1]: Leaving directory `/root/openh323/src' make: *** [opt] Error 2 [EMAIL PROTECTED] openh323]# - Now Asterisk exit with this error when I try to start it: -- more stuff... [cdr_pgsql.so] = (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found [chan_h323.so]Feb 16 13:02:21 WARNING[31879]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Feb 16 13:02:21 WARNING[31879]: loader.c:440 load_modules: Loading module chan_h323.so failed! [EMAIL PROTECTED] asterisk-1.0.5]# My system: Red Hat v.9.0 What can I do ? Thanks. ___ 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] SIP hassels
hi all i have created an accoutn with sip.phonehome.co.za and register it in asterisk this seems to have no problem as sip show peers displays the connection to the sip proxy but when i make a call from an extension to the sip number after dialing the phone starts ringing immidiately a tcpdump show the server communicating with the sip proxy over the net 1. but an error pops up in * chan_local.c:389 local_Alloc: No such extension/context [EMAIL PROTECTED] creating local channel 2. app_dial.c:356 wait_for_answer: unabvle to create local channel for call forward to 'Local/[EMAIL PROTECTED]' (cause = 0) i use sip.phonehome.co.za my number with them is 8101321 any help appreciated ___ 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] Extra sounds (Weather)
Liaan vd Merwe wrote on Wednesday, 16 February 2005 2:53 AM: This is the example script (extracted from that link) you will need to find a weather page for your region an then change the urls and grep statements chow L Once again, this is NOT the script mentioned at Eric Wieling's former site, http://www.fnords.org/~eric/asterisk/, referenced it the message in the archives at http://lists.digium.com/pipermail/asterisk-users/2003-November/025983.html. Sincerely, Trevor Hammonds ___ 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] Extra sounds (Weather)
Hi Trevor This i know I just send you a other script doing the same task this will give you a guideline to make you own - Original Message - From: Trevor G. Hammonds [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 1:50 PM Subject: RE: [Asterisk-Users] Extra sounds (Weather) Liaan vd Merwe wrote on Wednesday, 16 February 2005 2:53 AM: This is the example script (extracted from that link) you will need to find a weather page for your region an then change the urls and grep statements chow L Once again, this is NOT the script mentioned at Eric Wieling's former site, http://www.fnords.org/~eric/asterisk/, referenced it the message in the archives at http://lists.digium.com/pipermail/asterisk-users/2003-November/025983.html. Sincerely, Trevor Hammonds ___ 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] Extra sounds (Weather)
On Wed, 16 Feb 2005 03:50:28 -0800, Trevor G. Hammonds [EMAIL PROTECTED] wrote: Liaan vd Merwe wrote on Wednesday, 16 February 2005 2:53 AM: This is the example script (extracted from that link) you will need to find a weather page for your region an then change the urls and grep statements chow L Once again, this is NOT the script mentioned at Eric Wieling's former site, http://www.fnords.org/~eric/asterisk/, referenced it the message in the archives at http://lists.digium.com/pipermail/asterisk-users/2003-November/025983.html. ... and nor is it what the OP was asking for - a script to use the pre-recorded weather terms in the loligo.com extra sounds package :-) --- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ 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] Dial (Local/.....)
Hi ALL; I saw several examples of "Dial" app with the format: Dial(Local/..) Anybody knows what the "Local" technology means? Regards Mohamamd ___ 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] Asterisk exist with error
On Wednesday 16 February 2005 11:40, Nemesis wrote: Hello ! [...] ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Feb 16 13:02:21 WARNING[31879]: loader.c:440 load_modules: Loading module chan_h323.so failed! [EMAIL PROTECTED] asterisk-1.0.5]# My system: Red Hat v.9.0 What can I do ? Use google. ___ 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] Dial (Local/.....)
On Wed, 16 Feb 2005 03:40:38 +0330, mohammad [EMAIL PROTECTED] wrote: I saw several examples of Dial app with the format: Dial(Local/..) Anybody knows what the Local technology means? Did you try the WiKi? Or Google? http://www.google.com/search?q=asterisk+local -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ 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] chan_sip errors on CVS HEAD
I've got a test * server (hppbx) where I install CVS-HEAD as often as possible, with my extension registered to this, talking through IAX to our production server which then channels out to the PSTN. After completing a call just now, the following appeared on the CLI of hppbx (the 90xxx is a valid number, changed to protect the guilty): == Spawn extension (from-sip, 90xxx, 1) exited non-zero on 'SIP/711-31db' Feb 16 12:42:38 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:39 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:41 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:45 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:49 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:53 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:57 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy hppbx*CLI show version Asterisk CVS-HEAD-02/05/05-09:30:42 built by [EMAIL PROTECTED] on a i686 running Linux Feb 16 12:43:01 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:43:05 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:43:09 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy hppbx*CLI show channels Channel (ContextExtensionPri ) State Appl. Data 0 active channel(s) There are no more errors after this. Anyone else had this ? Julian. ___ 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] Ser 0.9.0 adding a user?
LOL, I'm a dumba$$ please ignore :-) -Original Message- From: Matt Schulte Sent: Tuesday, February 15, 2005 2:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Ser 0.9.0 adding a user? I get this when adding a user in ser (using serctl) [EMAIL PROTECTED] sbin]# ./serctl add +18165551212 blahblah [EMAIL PROTECTED] MySql password: error: 400; check if you use aliases in SER Um error 400?? I'm lost. no docs, frustrated. venting. Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ser 0.9.0 adding a user?
Matt Schulte wrote: LOL, I'm a dumba$$ please ignore :-) Might help to post what you did wrong for the archives...although, I guess it isn't really Asterisk related. :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Dialplan + Registrar DB
Hi; As you probably know, SER style of handling an incoming call is : 1) try to look-up it from registrar DB 2) if not found there, try to do some thing else Is there any possibility of doingthe aboveat "Asterisk Dial-plan"? Regards Mohammad ___ 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] ATA's
David Uzzell wrote: Shaun Ewing wrote: I'm using the ATA 186 and think it's great. The latest firmware also changes the web interface - it's similar to the 7905G/7912G phones now. Did you have much trouble getting the ATA 186 working? No. I have one running Version: v3.1.0 atasip (Build 040211A) I'm using Version: v3.0.0 atasip (Build 031210A). I have it setup and it does poll the * server but does not work to use and errors in sip. Followed the instructions on the wiki page for them and it still wants to be a pain :( Other problem is that it is in Denmark and I am in AUS :) so timming is an issue. Any advice would be appreciated. Looks like you have a NAT problem. You need to be more specific about your NAT setup. ___ 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] Asterisk exist with error
At 12:10 16-02-05 +, you wrote: On Wednesday 16 February 2005 11:40, Nemesis wrote: Hello ! [...] ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Feb 16 13:02:21 WARNING[31879]: loader.c:440 load_modules: Loading module chan_h323.so failed! [EMAIL PROTECTED] asterisk-1.0.5]# My system: Red Hat v.9.0 What can I do ? Use google. Found nothing to solve this problem there :( So if anybody now ___ 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] Dialplan + Registrar DB
mohammad wrote: Hi; As you probably know, SER style of handling an incoming call is : 1) try to look-up it from registrar DB 2) if not found there, try to do some thing else Is there any possibility of doing the above at Asterisk Dial-plan? Just forward the call to Asterisk if it has a certain URI. I.E. sip address starts with 7,8 or 9 then send to Asterisk. Then you can do whatever you like with the call in Asterisk. I.E. I have features on 7 (i.e. 700=voicemail), iax extensions etc on 8, and 9 for outgoing calls. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Dial (Local/.....)
Mondial Software Limited 020 7043 2795 www.mondialsoftware.com Click here to view our presentation of Cash Controller showing its forecasting and automated bank reconciliation features -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: February 16, 2005 12:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dial (Local/.) On Wed, 16 Feb 2005 03:40:38 +0330, mohammad [EMAIL PROTECTED] wrote: I saw several examples of Dial app with the format: Dial(Local/..) Anybody knows what the Local technology means? Did you try the WiKi? Or Google? http://www.google.com/search?q=asterisk+local -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ 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
[Asterisk-Users] MARK: Sip No inbound audio
when connecting through a sip proxy server outside our network on the net to connect to a land line. the call connects to the land line. but we cannot hear the other party they can hear us. ___ 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] Asterisk exist with error
On Wed, 16 Feb 2005 15:42:18 +0200, Nemesis [EMAIL PROTECTED] wrote: At 12:10 16-02-05 +, you wrote: On Wednesday 16 February 2005 11:40, Nemesis wrote: Hello ! [...] ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Feb 16 13:02:21 WARNING[31879]: loader.c:440 load_modules: Loading module chan_h323.so failed! [EMAIL PROTECTED] asterisk-1.0.5]# My system: Red Hat v.9.0 What can I do ? Use google. Found nothing to solve this problem there :( So if anybody now I just pasted this line from your error message into Google: ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object and the top result looks to have some good advice for you. Did you try that? Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ 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] Strict Routing vs Loose Routing
Please file a bug report at bugs.digium.com - thanks! Cheers, Philipp ___ 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] DTMF inband detection improvement
Hi Florian, If you really are using ulaw, and you do not have extreme packet loss or jitter, DTMF detection should be very reliable. It is no better in CVS HEAD because it wasn't broken in the first place. What does wrong with DTMF detection? Do you realise how DTMF from a GSM phone works? If you get digits when you press them really slowly on the phone, but miss some when you press them fast, then welcome to the land of GSM :-) The DTMF tones don't come from phone. The come from the basestation. Most basestations stretch each digit to well over a second. It really screws up any attempt at timing based entry methods. :-( Regards, Steve Florian Lefeuvre wrote: Hi all, I have some probleem detecting DTMF send by a GSM phone, I'm using SIP with ulaw. do you know what are the options to improve the detection ? I'm using asterisk 1.05, is the CVS HEAD version had some improvement about DTMF detection? Florian. ___ 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] iax.cc and/or Sixtel.net seems like IT IS A SCAM.
On Tue, Feb 15, 2005 at 07:42:56PM -0500, Nabeel Jafferali said: 4. Scnet.net has 5 pages website (quite a work for ISP), that any kid could create in 1h scnet.net is Server Central, a data centre where my host (HostForWeb), among others, maintains their servers. I do know it is a reliable data centre and I doubt is in any way related to iax.cc and/or sixtel.net other than housing their server(s). Yep. I've used server central as well. They DO have a reliable network, and hosting centers in several cities (including San Jose at Equinix.) ___ 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] Ser 0.9.0 adding a user?
What I did wrong was post it to the wrong list, heheh *shame on me* and no, still no resolution. :-( -Original Message- From: Matt Riddell [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 16, 2005 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Ser 0.9.0 adding a user? Matt Schulte wrote: LOL, I'm a dumba$$ please ignore :-) Might help to post what you did wrong for the archives...although, I guess it isn't really Asterisk related. :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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
[Asterisk-Users] ZAP channel on TE410P doesn't hang up
Hello * users I've have a rather disturbing problem, which I don't know how to debug or how to solve, but first a brief description of the set up. One Asterisk server with a TE410P card installed (first line used on this only), and a number of Wellgate 3504A (4 port FXS devices with SIP firmware). There is no connection from the Asterisk server to the outside world or any other IPTEL providers. The server only acts as a PABX and PSTN gateway for the SIP devices. Now the problem; sometimes the ZAP line isn't disconnected properly, I don't know what causes this and haven't been able to reproduce this behaviour, which is why I haven't got a clue how to debug the problem. The way I came across this issue was by examining CDR files from our telco provider, which showed that some calls had been "hanging" for over 24 hours. The "funny" thing is that the major part of these hanging calls was to another Asterisk server PSTN-PSTN (Almost same set up) and as far as I've been able to interpret the calls have been answered by Asterisk Voicemail (I don't know if this is of any importance). Have anybody experienced the same behaviour or got any ideas to what can be done, as this gets rather expensive over time. I've googled a lot to find any clues, but only found some similar problems on the X100P board. As a side note I've now implemented an AbsoluteTimeout for the call, I know this isn't a solution but merely a workaround. Please let me know which configs or logs to provide, any help is greatly appreciated. Kind regards, Mickey Binder ___ 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] Attended xfer
I am running 1.0.5 and can happily blind xfer from extension to extension, but I can't blind xfer. I have read various snippets about #2 or #8 or other such key combos, but nothing seems to let me do attended xfer. From xlite I can blind xfer without problem but no attended xfer. For attendant transfer you should use CVS Head, in Asterisk stable is not implemented that feature! ___ 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: PSTN incoming - both SIP H323 always arrive in default context :-?
I'm seeing the same problem here, all SIP calls go to the default context. Kelvin Chua wrote: this is something i just recently noticed. have you found any info on how to manage incoming calls through chan_h323? it doesn't seem to match any entity you define, it always uses the default context... On Sat, 2004-01-24 at 02:39, Fran Boon wrote: Some of you may remember seeing my issue using SIP for incoming calls from the PSTN: http://voip-info.org/wiki-Asterisk+cisco+FXO i.e. all incoming calls arrive in the default 'bogon-calls' context. Well, I tried again using H.323 get exactly the same result (both for chan_h323 chan_oh323) i.e. all attempts to put a type=peer in sip.conf or a type=user in h323.conf for my host are ignored/bypassed. Is this a bug? Luckily for me, I can firewall off the H.323 port to all bar this one IP, so I now have a workable solution...until I want to extend the H.323 gateway to other devices... Anyone get host=x.x.x.x to be able to bypass the default contexts with either SIP or H.323? Cheers, Fran. ___ 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 ___ 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] Which IP phone to use in Australia
On Wed, 16 Feb 2005 22:04:27 +1100, Adam Goryachev [EMAIL PROTECTED] wrote: BTW, Polycom *don't* say you can't use their phone in/with any particular manner/software. All they say is that if the phone breaks, and it is caused by asterisk, then they won't really help you out. However, the fact that it works, and works well pretty much says it all. In my experience (I've had very limited experience with the cisco phones) the polycom phone is better than the cisco. They are equal in most ways, except the cisco phones require you to pay some silly licensing fee, and if you buy the phone second hand, then you can't use any firmware version without purchasing it extra At least polycom provide the software to download. Interesting. In that case, I'll bite. Anybody know of a place in Australia that sells them (preferably online)? I might look into getting a small quantity (1 or 2) to get acquanted with them. -Shaun Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au -Shaun ___ 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] Dial (Local/.....)
Are you paying me? Did I ask you to do this? Did you get permission from all 10,000 to do this? On Wed, 16 Feb 2005 13:40:41 -, Bill Seddon [EMAIL PROTECTED] wrote: Mondial Software Limited 020 7043 2795 www.mondialsoftware.com Click here to view our presentation of Cash Controller showing its forecasting and automated bank reconciliation features ___ 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] Dialplan + Registrar DB
Matt Riddell wrote: mohammad wrote: As you probably know, SER style of handling an incoming call is : 1) try to look-up it from registrar DB 2) if not found there, try to do some thing else Is there any possibility of doing the above at Asterisk Dial-plan? Just forward the call to Asterisk if it has a certain URI. I.E. sip address starts with 7,8 or 9 then send to Asterisk. Then you can do whatever you like with the call in Asterisk. I.E. I have features on 7 (i.e. 700=voicemail), iax extensions etc on 8, and 9 for outgoing calls. Is that what you were looking for? If not, can you explain this registrar db concept you're talking about? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Attended xfer
CVS in a production environment? Is that advisable? [EMAIL PROTECTED] wrote: I am running 1.0.5 and can happily blind xfer from extension to extension, but I can't blind xfer. I have read various snippets about #2 or #8 or other such key combos, but nothing seems to let me do attended xfer. From xlite I can blind xfer without problem but no attended xfer. For attendant transfer you should use CVS Head, in Asterisk stable is not implemented that feature! ___ 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
[Asterisk-Users] ZAP channel on TE410P doesn't hang up (Plain Text this time)
Hello * users Sorry I forgot to send the mail in plain text the first time... I've have a rather disturbing problem, which I don't know how to debug or how to solve, but first a brief description of the set up. One Asterisk server with a TE410P card installed (first line used on this only), and a number of Wellgate 3504A (4 port FXS devices with SIP firmware). There is no connection from the Asterisk server to the outside world or any other IPTEL providers. The server only acts as a PABX and PSTN gateway for the SIP devices. Now the problem; sometimes the ZAP line isn't disconnected properly, I don't know what causes this and haven't been able to reproduce this behaviour, which is why I haven't got a clue how to debug the problem. The way I came across this issue was by examining CDR files from our telco provider, which showed that some calls had been hanging for over 24 hours. The funny thing is that the major part of these hanging calls was to another Asterisk server PSTN-PSTN (Almost same set up) and as far as I've been able to interpret the calls have been answered by Asterisk Voicemail (I don't know if this is of any importance). Have anybody experienced the same behaviour or got any ideas to what can be done, as this gets rather expensive over time. I've googled a lot to find any clues, but only found some similar problems on the X100P board. As a side note I've now implemented an AbsoluteTimeout for the call, I know this isn't a solution but merely a workaround. Please let me know which configs or logs to provide, any help is greatly appreciated. Kind regards, Mickey Binder ___ 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] iax.cc and/or Sixtel.net seems like IT IS A SCAM.
I have an 866# number with iax.cc. It works fine. It did take me a couple of days to get it and they did have some problem the first day it was active, but I was able to contact support via IM and email. They resolved the problems and the service is working fine for me. Although, I still haven't gotten the 734 DID that I ordered on 1-28. (The 866 number was more important to me, so I don't mind the delays on the 734 DID so much.) Is IAX.cc / Sixtel.net a scam? No. Do they have provisioning problems? Yes. Does the service work once you have the numbers? Yes. cheers, glenn ___ 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] Asterisk, inband DTMF send by a GSM mobile
On Feb 15, 2005, at 1:12 PM, Florian Lefeuvre wrote: I find the compilation option RADIO_RELAX. this option change a threshold in DTMF detection (function dtmf_detect in dsp.c) I remark an big improvement in the detection of the dtmf over GSM. have you ever test this option? RADIO is obscur for me, does it mean all wireless device? I haven't dug into the source looking to fix this problem so, no, I haven't tried enabling that option. Perhaps someone on the list knows more about this option? I'll certainly try it out as well. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] asterisk ipv6
Hi Has anybody tried using asterisk on an ipv6 internal network? If so, any feedback or comments would be very appreciated. Im not sure, but is ipv6 a real-time protocol already? ,jm ___ 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] solid-state asterisk pbx?
On Tue, 15 Feb 2005 14:05:36 +0100, Vledder, Hans wrote: I've been thinking of making a (mostly) solid-state asterisk pbx. Take either centos or some other distro, cut it down to bare minimum and put asterisk + AMP on. Something that could be put onto a usb2.0 flash stick, bootable. Modern flash devices (usb, compactflash) have builtin wear leveling management and will last longer than you think: http://www.sandisk.com/pdf/oem/WPaperWearLevelv1.0.pdf Use ramdisk to store temporary files and flash to store permanent pbx configuration data, voicemail etc. Done right, one could literally have a pbx on a stick. Eg a 256mb, 512mb or 1gb sandisk usb2.0 dongle. Anyone done something like this yet? Andy Powell has prepared a CF image at www.automated.it/asterisk. I have been able to get this booted on a testbed system. Sadly, I'm a Linux newbie and not skilled at command line administration, thus I'm stuck at the moment. I can get the existing image running, but have not been able to get ssh working, change passwords, load my configs to the CF, etc. If there's someone on-list who could assist in this regard I'd gladly share my experience moving my production server to be CF based. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] DTMF inband detection improvement
On Wed, 16 Feb 2005, Steve Underwood wrote: If you really are using ulaw, and you do not have extreme packet loss or jitter, DTMF detection should be very reliable. It is no better in CVS HEAD because it wasn't broken in the first place. We have some problems with dtmf detection on our lines. We use E1 pris connected to a TE405P. Mostly we see duplicated digits. Unless the signal is perfect and distortion free Asterisk sees the small imperfections as the end of the digit. We can provoke this problem from some of our office phones (when on speaker phone). Asterisk is more or less the only place we see this problem. Peter ___ 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] ATA that actually work with T.38
On Feb 15, 2005, at 2:32 PM, Matthew Boehm wrote: Stop. The PAP2-NA's have no T38 support. Next time, lets try and read the OP's message before responding. -Matthew Hah! With over 2-- or 300 messages per day we're supposed to read everything in them? I find it easier just to respond to multiple posts in long response anyhow! ;-) -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Attended xfer
On Wed, 2005-02-16 at 14:11 +, Mark Benson wrote: As for the alternative to attended xfer, parking calls, I'm guessing this is just a case of blind xfering calls to a parking extension? That is correct... if 800 is your parking extension then you dial #800, you will hear what extension they were parked on (i.e. 801, 802, etc...) and they will hear MOH. Then call your party and tell them what extension the call is parked on. This is all much easier if you use phones with programmable buttons. Just set up Park, Park 1, Park 2, etc... buttons on all the handsets and nobody has to remember anything. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] HELP!!!!!!!!
Hi, I have installed two X-Lite phones and theyre able to login successfully. The two phones plus the Asterisk system are all on the same LAN with private addresses assigned to each of them. When a call is initiated and is picked up on the other end, there is completely no sound at all (as in the line goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX. From the Asterisk CLI I see the following errors; i) Unknown RTP codec 72 received ii) RFC3389 support incomplete Anyone got ideas on how I can go about this? Thanks in advance. Julius Kidubuka When you do the common things in life in an uncommon way, you will command the attention of the world ___ 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] Attended xfer
Does anyone know if the attended transfer in CVS head works with app_queue (and more importantly, chan_agent ?) This is the only thing stopping me from deploying the attended transfer patches. Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Benson Sent: 16 February 2005 14:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Attended xfer CVS in a production environment? Is that advisable? [EMAIL PROTECTED] wrote: I am running 1.0.5 and can happily blind xfer from extension to extension, but I can't blind xfer. I have read various snippets about #2 or #8 or other such key combos, but nothing seems to let me do attended xfer. From xlite I can blind xfer without problem but no attended xfer. For attendant transfer you should use CVS Head, in Asterisk stable is not implemented that feature! ___ 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 ___ 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] ATA's
On Feb 15, 2005, at 6:08 PM, Leo Ann Boon wrote: From my experience, the ATA is a very solid, dependable piece of hardware. I was told by a source in the company that OEMs for Cisco, the units are expensive because of the high quality parts being used. The web config looks crappy but otherwise where else do you find a $100 device that does SCCP/MGCP/SIP/H323? None of the competitors even come close to that level of protocol support. For developers who have to work on various protocols, the ATA is really cool. Guess I never really looked at it that way. Perhaps when if I cancel my Vonage account I'll just hang on to the ATA [he casually comments as the collection of VOIP adapters steadily grows in the basement...]. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] DTMF inband detection improvement
On Feb 16, 2005, at 10:01 AM, Peter Svensson wrote: On Wed, 16 Feb 2005, Steve Underwood wrote: If you really are using ulaw, and you do not have extreme packet loss or jitter, DTMF detection should be very reliable. It is no better in CVS HEAD because it wasn't broken in the first place. We have some problems with dtmf detection on our lines. We use E1 pris connected to a TE405P. Mostly we see duplicated digits. Unless the signal is perfect and distortion free Asterisk sees the small imperfections as the end of the digit. We can provoke this problem from some of our office phones (when on speaker phone). Asterisk is more or less the only place we see this problem. I concur. I have DTMF problems with inbound calls over IAX. Don't have any DTMF problems locally using g.711. I also have problems with inbound calls from GSM phones but hey, that's not a surprise and yes, if you dial realy slow then it seems to work more reliably. BTW, Steve, if you're still reading, what is the RADIO_RELAX option intended to be for in dsp.c? -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] DTMF inband detection improvement
Hi Peter, If that is true, someone must have broken something. Not only does the DTMF detector I wrote not care about small imperfections, it even tolerates a dropped packet with the DTMF passes over a VoIP path (this kind of tolerance was added a couple of years ago). Regards, Steve Peter Svensson wrote: We have some problems with dtmf detection on our lines. We use E1 pris connected to a TE405P. Mostly we see duplicated digits. Unless the signal is perfect and distortion free Asterisk sees the small imperfections as the end of the digit. We can provoke this problem from some of our office phones (when on speaker phone). Asterisk is more or less the only place we see this problem. Peter ___ 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] DTMF inband detection improvement
Hi Mark, Mark Eissler wrote: BTW, Steve, if you're still reading, what is the RADIO_RELAX option intended to be for in dsp.c? It is something someone else added to the code to make the detection criteria in relaxed mode even more relaxed. If setting that helps, something in your channel must be causing some serious filtering of low frequencies. Can you try logging the audio to a file, and send it to me for analysis? chan_spy, or something like that, should do the job. Regards, Steve ___ 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] DTMF inband detection improvement
Steve: Given this and the number of recent messages related to DTMF problems can you add any thoughts on how to improve implementations? I too have problems correctly handling DTMF in my environment. In the LAN /IP world DTMF works as expected to allow an IP phone user to access my Asterisk based menu. Calls from the PSTN usually work as expected but occasionally someone reports that pressing the keypad digits has no affect. Instead the timeout clause is eventually hit and the caller is disconnected. Cell phone users cannot even use their phone keypad to respond to prompts. They always timeout. Thanks,Steve Steve Underwood wrote: Hi Peter, If that is true, someone must have broken something. Not only does the DTMF detector I wrote not care about small imperfections, it even tolerates a dropped packet with the DTMF passes over a VoIP path (this kind of tolerance was added a couple of years ago). Regards, Steve Peter Svensson wrote: We have some problems with dtmf detection on our lines. We use E1 pris connected to a TE405P. Mostly we see duplicated digits. Unless the signal is perfect and distortion free Asterisk sees the small imperfections as the end of the digit. We can provoke this problem from some of our office phones (when on speaker phone). Asterisk is more or less the only place we see this problem. Peter ___ 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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ 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] Help With Broadvoice {Scanned}
Here is my conf files. sip.conf register = phone#:sip/[EMAIL PROTECTED] type=friend username=phone# fromuser=phone# secret=sip/passwd host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice1 dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=no [bv-in-1] type=friend host=sip.broadvoice.com context=from-broadvoice1 dtmfmode=inband canreinvite=no nat=no allow=ulaw extensions.conf exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _011,1,Dial(SIP/[EMAIL PROTECTED]) On Wed, 2005-02-16 at 00:52 -0500, Randy Johnson wrote: Remember that the password is not your broadvoice website password but the one you need to get from broadvoice support. Randy Greg Hill wrote: On Tue, 15 Feb 2005, Max Clark wrote: I have experimented with several configs based on different pages and threads but nothing is working. How do I properly configure my broadvoice account? [general] register = [EMAIL PROTECTED]:pass:[EMAIL PROTECTED] the register I'm using looks like this: register = 310584:pass@sip.broadvoice.com [broadvoice] type=peer host=sip.broadvoice.com secret=pass fromuser=310584 fromdomain=sip.broadvoice.com context=incoming dtmfmode=inband canreinvite=no nat=yes qualify=yes try: [broadvoice] type=peer username=310584 secret=pass host=sip.broadvoice.com port=5060 context=incoming fromuser=310584 fromdomain=sip.broadvoice.com dtmfmode=inband canreinvite=no insecure=very permit=147.135.8.128/32 qualify=yes and adjust your permit= line to match the IP of the BV proxy you've set in your /etc/hosts (proxy.[chi|lax|dca|one other I forgot].broadvoice.com). Try a blend of this stuff with whatever the most recent recommendation on their support page says. Greg ___ 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 -- David Shaw [EMAIL PROTECTED] ___ 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] HELP!!!!!!!!
Yes turn off silence suppression. xlite - Menu - Advanced - audio settings - Silence Settings - transmite Silence: (change to yes) - Original Message - From: Julius Kidubuka To: asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 10:04 AM Subject: [Asterisk-Users] HELP Hi, I have installed two X-Lite phones and theyre able to login successfully. The two phones plus the Asterisk system are all on the same LAN with private addresses assigned to each of them. When a call is initiated and is picked up on the other end, there is completely no sound at all (as in the line goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX. From the Asterisk CLI I see the following errors; i) Unknown RTP codec 72 received ii) RFC3389 support incomplete Anyone got ideas on how I can go about this? Thanks in advance. Julius Kidubuka "When you do the common things in life in an uncommon way, you will command the attention of the world" ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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
[Asterisk-Users] Why Asterisk can't cope with silence suppression?
OK I have to ask. Why is it that Asterisk can't cope with silence suppression? All the clients seem to be able to but not Asterisk. What would be needed to get it to work with silence suppression? What is the problem? ___ 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] Dutch VOIP-PSTN provider
Hi, I read a lot about US providers that can terminate a PSTN number for you and offer IAX or SIP connectivity. Does anyone know such a company in The Netherlands ? I read about Unet. Anyone with experience with them ? Any information is welcome. -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ 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] Monitor does not like variable subsitutions
Hello,
I have been attempting to get the Monitor function to
accept a loal variable substitution in order to use
the same filename later in the same context. Monitor
does not appear to like it, as it attempts to use
wav|filename as the recording type, as opposed to just
wav.
Here is what I get if I just supply a filename
directly (it works fine):
--context-
exten = _9X.,3,Monitor(wav|recording|m)
--context-
--CLI-
-- Executing SetVar(SIP/3004-275c,
REC_FILE_NAME=rec_to_448704386865_at_16022005-16:54:10)
in new stack
-- Executing Monitor(SIP/3004-275c,
wav|recording|m) in new stack
-- Executing AGI(SIP/3004-275c, outbound.agi)
in new stack
--CLI-
Here is what I get when I attempt to to variable
substituion for the filename:
--context-
exten =
_9X.,2,SetVar(REC_FILE_NAME=rec_to_${EXTEN:1}_at_${DATETIME})
exten = _9X.,3,Monitor(wav|${FILENAME}|m)
--context-
--CLI-
-- Executing SetVar(SIP/3004-da21,
REC_FILE_NAME=rec_to_448704386865_at_16022005-16:56:35)
in new stack
-- Executing Monitor(SIP/3004-da21,
wav|rec_to_448704386865_at_16022005-16:56:35|m) in
new stack
Feb 16 16:56:35 WARNING[17028]: file.c:934
ast_writefile: No such format
'wav|rec_to_448704386865_at_16022005-16'
Feb 16 16:56:35 WARNING[17028]: res_monitor.c:154
ast_monitor_start: Could not create file
/var/spool/asterisk/monitor/m-in
Feb 16 16:56:35 WARNING[17028]: res_monitor.c:300
ast_monitor_change_fname: Cannot change monitor
filename of channel SIP/3004-da21 to m, monitoring not
started-- Executing AGI(SIP/3004-da21,
outbound.agi) in new stack
--CLI-
I do believe that I had this working before (I am
running the CVS HEAD from yesterday).
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[Asterisk-Users] Can't connect Snom 190 to Asterix PBX. Suggestions?
Hello, I'm attempting to get Asterisk running for the first time in my company. As I've never used it before, I am creating a small testbed with which to learn Asterisk and get the kinks worked out before attempting to roll it out. I have * compiled and running, and built the sample config files as suggested by the Wiki. I got my Snom 190 configured to use DHCP, and have created an entry in my sip.conf for the phone. However, when * boots, I get the following message: -- Got SIP response 404 Not Found back from 192.168.2.60 When I look at the SIP trace logs in the phone, I see the following: Received from udp:192.168.2.15:5060 at 16/2/2005 09:38:54:330 (463 bytes): NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK5e3acadf From: asterisk sip:[EMAIL PROTECTED];tag=as65dd26cd To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/0 Sent to udp:192.168.2.15:5060 at 16/2/2005 09:38:54:330 (267 bytes): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK5e3acadf From: asterisk sip:[EMAIL PROTECTED];tag=as65dd26cd To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY Content-Length: 0 The phone is a Snom 190, running firmware snom190-3.56q-SIP-j.bin (from what I can tell). The only Asterisk configuration I have done was in sip.conf. I added the following entry for the phone: [test1] type=friend; Friends place calls and receive calls context=from-sip ; Context for incoming calls from this user secret=blah language=en; Use German prompts for this user host=dynamic ; This peer register with us dtmfmode=inband; Choices are inband, rfc2833, or info defaultip=192.168.2.60 ; IP used until peer registers username=snom ; Username to use in INVITE until peer registers mailbox=1234,2345 ; Mailboxes for message waiting indicator restrictcid=yes; To have the callerid restriced - sent as ANI disallow=all allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! [EMAIL PROTECTED],2345 ; Mailbox(-es) for message waiting indicator The IP of the Snom is indeed 192.168.2.60. Asterisk runs at 192.168.2.15. I'm using the latest stable version of *, and it's running on FreeBSD 5.3. I did not build from the ports tree. What am I missing? When the phone is configured, how do I verify that it's working? Any help is greatly appreciated! Thanks in advance, Jason -- Jason A. Crome Senior Software Engineer, DEVNET, Inc. E-Mail: [EMAIL PROTECTED] http://www.devnetinc.com ___ 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] Why Asterisk can't cope with silence suppression?
OK I have to ask. Why is it that Asterisk can't cope with silence suppression? All the clients seem to be able to but not Asterisk. What would be needed to get it to work with silence suppression? What is the problem? Essentially its because * has been architected to send an rtp packet after receiving a packet. If * never see's and incoming rtp packet, then it won't send an rtp packet (which usually contains some amount of audio). Thus choppy audio in one direction. ___ 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: How do I match a D? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
-Original Message- From: Chris Wade [mailto:[EMAIL PROTECTED] Brian Roy wrote: I think that my PBX does this too. Is there any way I can get the Zaptel drivers to disconnect on that tone too? I would love to replace my existing voicemail with * but I can't get my PBX to signal a disconnect properly. I have to use busycount=10 but every voicemail has an annoying busy signal tacked onto the end of it. CVS HEAD has a features.conf entry for 'disconnect' which is normally set to '*', you might try placing a 'D' there and see if that works? I just played with that a little, but it doesn't seem to do anything as far as voicemail is concerned. Maybe I'm missing something. It'd be really nice if there were a way to get voicemail (and other apps, like Directory()) to properly interpret this. My phone system doesn't give a busy after a hangup, just a 'D' followed by silence, so right now I'm coping by setting maxsilence=5 and review=no in my voicemail.conf. Not ideal, but it works. I've considered trying to hack the voicemail source code to handle the 'D', but I haven't really dug into it. ___ 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] DTMF inband detection improvement
On Wed, 16 Feb 2005, Steve Underwood wrote: If that is true, someone must have broken something. Not only does the DTMF detector I wrote not care about small imperfections, it even tolerates a dropped packet with the DTMF passes over a VoIP path (this kind of tolerance was added a couple of years ago). Actually, the problem has gone away for us, probably after some upgrade. We are running a cvs head version from december. The phones that used to be problematic no longer are. The last time we seem to have seen that problem according to the logs was in september. After an upgrade there the problems went away. Asterisk does handle imperfect DTMF tones correctly for us on our pure TDM setup. We use VoIP very little and only over a lightly loaded lan and have not noted any problems there either. Sorry about the false alarm. Peter ___ 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] chan_misdn and hylafax
Hi, do you need a sending or receiving fax solution ? Receiving fax via Asterisk and misdn - no problem, but i have no sending fax solution at this time. Andreas. --On Freitag, 11. Februar 2005 15:04 + Anabela Abreu [EMAIL PROTECTED] wrote: Was anyone put hylafax working with chan_misdn? ___ 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] Passthrough and reInvite
Tom Samplonius wrote: It is not clear how exactly g729 pass-through can be enabled. I have a SIP call off a gateway come into an Asterisk menu, and then I send the SIP call to another SIP gateway using Dial(). Even though codec preferences have g729 listed first, it never gets used. Without actually seeing your config files it's hard to guess as to why that might be. Also keep in mind that if you answer the call in the dialplan and want to play messages, you will either need those messages already formatted as G.729 files or G.729 encoder licenses to be able to play them to the caller. Both gateways have separate peer entries in sip.conf, and both have canreinvite=yes set. Can Asterisk change the media type during a re-Invite? The call is answered as g711u initially, and then Asterisk plays a menu, and then does a Dial(). I can see Asterisk doing the reInvite, but the protocol stays at g711u. No, Asterisk never changes the codec once the call is established, even when redirecting the media elsewhere. ___ 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] Why Asterisk can't cope with silence suppression?
On Wed, 16 Feb 2005, Rob Scott wrote: Why is it that Asterisk can't cope with silence suppression? All the clients seem to be able to but not Asterisk. What would be needed to get it to work with silence suppression? What is the problem? Asterisk clocks outgoing rtp data to a device from the incoming rtp stream from the same device. This is a known limitation and there has been some talk about implementing an internal clocking system. In addition, Asterisk should generate comfort noise when the rtp stream is quiet due to silence supression (which is signalled with a CN packet). Perhaps the new jitter buffer will be able to handle this? Peter ___ 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] Can't connect Snom 190 to Asterix PBX. Sugge stions?
Here is a part of a working sip.conf for Asterisk with SNOM190 Phones. Try using host=dynamic and have a closer look at the configuration in the snom 190. Also, try using dtmfmode=rfc2833 . [general] realm = hallinux2.gwsnettech.local port = 5060 bindaddr = 0.0.0.0 context = default disallow=all allow=alaw allow=ulaw allow=gsm register = 081503:[EMAIL PROTECTED]/081503 language=de tos=0x04 [6301] type=friend username=6301 secret=xx host=dynamic disallow=all allow=alaw allow=ulaw allow=gsm dtmfmode=rfc2833 mailbox=6301 context=teilnehmer callgroup=1 pickupgroup=1 group=2 callerid = Guido Hecken [gwsNetTech] 6301 Hope, this helps... Guido Hecken -Ursprüngliche Nachricht- Von: Jason A. Crome [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 16. Februar 2005 17:05 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Can't connect Snom 190 to Asterix PBX. Suggestions? Hello, I'm attempting to get Asterisk running for the first time in my company. As I've never used it before, I am creating a small testbed with which to learn Asterisk and get the kinks worked out before attempting to roll it out. I have * compiled and running, and built the sample config files as suggested by the Wiki. I got my Snom 190 configured to use DHCP, and have created an entry in my sip.conf for the phone. However, when * boots, I get the following message: -- Got SIP response 404 Not Found back from 192.168.2.60 When I look at the SIP trace logs in the phone, I see the following: Received from udp:192.168.2.15:5060 at 16/2/2005 09:38:54:330 (463 bytes): NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK5e3acadf From: asterisk sip:[EMAIL PROTECTED];tag=as65dd26cd To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/0 Sent to udp:192.168.2.15:5060 at 16/2/2005 09:38:54:330 (267 bytes): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK5e3acadf From: asterisk sip:[EMAIL PROTECTED];tag=as65dd26cd To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY Content-Length: 0 The phone is a Snom 190, running firmware snom190-3.56q-SIP-j.bin (from what I can tell). The only Asterisk configuration I have done was in sip.conf. I added the following entry for the phone: [test1] type=friend; Friends place calls and receive calls context=from-sip ; Context for incoming calls from this user secret=blah language=en; Use German prompts for this user host=dynamic ; This peer register with us dtmfmode=inband; Choices are inband, rfc2833, or info defaultip=192.168.2.60 ; IP used until peer registers username=snom ; Username to use in INVITE until peer registers mailbox=1234,2345 ; Mailboxes for message waiting indicator restrictcid=yes; To have the callerid restriced - sent as ANI disallow=all allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! [EMAIL PROTECTED],2345 ; Mailbox(-es) for message waiting indicator The IP of the Snom is indeed 192.168.2.60. Asterisk runs at 192.168.2.15. I'm using the latest stable version of *, and it's running on FreeBSD 5.3. I did not build from the ports tree. What am I missing? When the phone is configured, how do I verify that it's working? Any help is greatly appreciated! Thanks in advance, Jason -- Jason A. Crome Senior Software Engineer, DEVNET, Inc. E-Mail: [EMAIL PROTECTED] http://www.devnetinc.com ___ 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
[Asterisk-Users] G729, NAT and Transcoding (all-in-one)
Got two phones here. 1 is Cisco 7960 and other is XTen Pro. Both have 729 capabilities and plenty of licenses on Asterisk. The Cisco phone has and registers/talks with asterisk on an internal IP (* = 10.0.3.10, phone = 10.0.3.151). The SIP peer for this phone is set to NAT=No and has this Codec Order: (g729|ulaw|alaw|gsm|g726). The XTen registers to the Asterisk external/public IP, even when it is inside our network. This SIP peer is set to NAT=yes and Codec Order: (g729) Both phones can (indepently) call PSTN numbers and talk fine. When XTen tries to call Cisco, there is a 1-way audio path. If you turn on G711 in XTen (but leave the SIP peer settings alone) then the call is fine. Firstly, why is asterisk even allowing this call when XTen's SIP peer has no 711 codec listing, and Second, why would a codec problem be doing this? -Matthew ___ 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] Using Hylafax and Digium T100P
My problem is that the 23 channels are going into asterisk. It seems that there is no way to pick off a couple of them and use them for faxes. Basically, I have 23 phone lines and can't use them for faxing. Thanks, Brian I am using a T100P for a 23-channel voice T1. Is it possible to create an extension that would allow sending a fax to HylaFax? Would I have the same problems as faxing through a TDM card? Can HylaFax send faxes through the T100P? Basically, is there any way to send/recieve faxes over one of the 23 T1 channels that * is currently using. If so, how? I have a channel bank (on a t100p) with a usr faxmodem plugged into one channel of it, and hylafax works fine with it. The odd fax machine has issues but I think it is the fax modem, not the channel bank, since I have a physical fax machine on another extension and it always works. The extension with the modem uses the asterisk fax autodetect from the main autoattendant structure, but if there is a problem I just tell people to call the extension with the real fax machine directly and hit start on their end. never had a problem with either of them outbound, you need to adjust your dialing prefixes etc, but that is the same for making any call. Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ 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] chan_misdn and hylafax
Yes i would like to have a solution with asterisk, hylafax and misdn. Em Wed, 16 Feb 2005 17:23:03 +0100 Andreas Czerniak [EMAIL PROTECTED] escreveu: Hi, do you need a sending or receiving fax solution ? Receiving fax via Asterisk and misdn - no problem, but i have no sending fax solution at this time. Andreas. --On Freitag, 11. Februar 2005 15:04 + Anabela Abreu [EMAIL PROTECTED] wrote: Was anyone put hylafax working with chan_misdn? ___ 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] Asterisk exist with error
At 13:41 16-02-05 +, you wrote: On Wed, 16 Feb 2005 15:42:18 +0200, Nemesis [EMAIL PROTECTED] wrote: At 12:10 16-02-05 +, you wrote: On Wednesday 16 February 2005 11:40, Nemesis wrote: Hello ! [...] ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Feb 16 13:02:21 WARNING[31879]: loader.c:440 load_modules: Loading module chan_h323.so failed! [EMAIL PROTECTED] asterisk-1.0.5]# My system: Red Hat v.9.0 What can I do ? Use google. Found nothing to solve this problem there :( So if anybody now I just pasted this line from your error message into Google: ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object and the top result looks to have some good advice for you. Did you try that? I browse http://lists.digium.com/pipermail/asterisk-users/2003-December/031368.html and tryed what is there...again, but with no result. PWLIBDIR=$HOME/pwlib export PWLIBDIR OPENH323DIR=$HOME/openh323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH I have discovert also that when I try: [EMAIL PROTECTED] /]# echo $LD_LIBRARY_PATH [EMAIL PROTECTED] /]# echo $PWLIBDIR [EMAIL PROTECTED] /]# echo $OPENH323DIR [EMAIL PROTECTED] /]# ...there is no answer... ___ 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] Using Hylafax and Digium T100P
On 2005.02.16 09:00 Brian M. Arlinghaus wrote: My problem is that the 23 channels are going into asterisk. It seems that there is no way to pick off a couple of them and use them for faxes. Basically, I have 23 phone lines and can't use them for faxing. You could always run another T100P into a HylaFAX-run T1 fax modem. That way you can use your T1 for faxing. Lee. ___ 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] [patch] fix libpri problem in Q931_INFORMATION handling
Hi there, I tried to use Voicemail from a PRI interface but it didn't work because pressing a key on the ISDN phone just caused Q931_IE_KEYPAD_FACILITY messages which are normally handled by a bri-stuffed libpri. Unfortunately a wrong if condition stops keypad messages from being passed to the channel driver. The patch attached to this mail removes the 2 lines from the code. Did I misunderstand something or is there really a bug? Deti --- q931.c~ 2005-02-16 18:21:33.907803750 +0100 +++ q931.c 2005-02-16 18:21:33.909803485 +0100 @@ -2877,8 +2877,7 @@ q931_release_complete(pri,c,PRI_CAUSE_INVALID_CALL_REFERENCE); break; } - if (c-ourcallstate!=Q931_CALL_STATE_OVERLAP_RECEIVING) - break; + pri-ev.e = PRI_EVENT_INFO_RECEIVED; pri-ev.ring.call = c; pri-ev.ring.channel = c-channelno | (c-ds1no 8); ___ 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] Using Hylafax and Digium T100P
On 2005.02.16 09:00 Brian M. Arlinghaus wrote: My problem is that the 23 channels are going into asterisk. It seems that there is no way to pick off a couple of them and use them for faxes. Basically, I have 23 phone lines and can't use them for faxing. You could always run another T100P into a HylaFAX-run T1 fax modem. That way you can use your T1 for faxing. you need a drop and insert channel bank. T1 from telco to channel bank, second T1 from channel bank to asterisk, analog ports on channel bank. by drop and insert you can either connect the analog ports to the telco t1 or the asterisk t1 and the rest of the channels pass through between the two t1's Lee. ___ 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 !DSPAM:42137ed031451817322028! Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ 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] speech recognition V 2.0
Greetings David, PerlBox would not be usable for the level of service that is needed by Asterisk to be viable Speech. PerlBox is a vocabulary based recognizer, or I as I call it a grunter, where you grunt something and it then does something cute. Grunters depend on you creating a vocabulary list that is different enough in the syllables so that it can tell cookie from kooky. So long as each grunt is different you can get a response from it. But a grunter can't do computer, call my mother-in-law in France using the PSTN connection. It can do Mom. If you talk to a computer the way you talk to a dog you don't get much more then sit, stay and down. The problem and the reason such Reco has never gained support is like all Voice/Speech Activated Dialing engines is that you have to remember all the vocabulary to use it. If you want to call Robert you can't say Bob. Grammar based recognition is the only solution. I know because I have watched several companies' which demanded that they can do it with grunter, which go out of business because the Stupid Customers won't learn how to use the system correctly. Maybe we should refer to the grunter engines as the Neanderthal engines. Not so much an evolutionary step so much as a evolutionary sidetrack experiment that went down a dead end. Grammar reco is not descendent from grunters; it starts further up the tree and is a distinct evolutionary line. As to PerlBox, I give kudos for their efforts for watch they have accomplished and their efforts are to be applauded for its segment. But it will not be able to progress into a useful and widely accepted Asterisk add-on. Sphinx and Festival are good projects. The last I worked with sphinx I was told that it would need modifications to make it more grammar aware, but that was 2 years ago and things may have improved. If not then Sphinx people please let me know when you will add grammars natively or refer me a grammar based engine. Race The Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David D. Faerman Sent: Tuesday, February 15, 2005 9:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] speech recognition V 2.0 hi seraching for info in the chat and in the web i found perlbox to meake speech recognition some one have any experience? any who to to put it to work? any help please thanks David ___ 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
[Asterisk-Users] Sphinx
Has anybody managed to implement Sphinx in their * system reasonably painlessly. if so: does it cause any problems with normal * operations. does it place any sort of constant heavy load on the machine. are there options for simple vs advanced implementations. all i am looking for is basicaly for a person to say a branch name. ie: johannesburg thanks ___ 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] Help With Broadvoice {Scanned}
David, Thanks for the reply. Just to clarify, is the register and first type=friend block all within the [general] section of sip.conf? Thanks, Max Max Clark max [at] clarksys.com http://www.clarksys.com David Shaw wrote: Here is my conf files. sip.conf register = phone#:sip/[EMAIL PROTECTED] type=friend username=phone# fromuser=phone# secret=sip/passwd host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice1 dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=no [bv-in-1] type=friend host=sip.broadvoice.com context=from-broadvoice1 dtmfmode=inband canreinvite=no nat=no allow=ulaw extensions.conf exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _011,1,Dial(SIP/[EMAIL PROTECTED]) On Wed, 2005-02-16 at 00:52 -0500, Randy Johnson wrote: Remember that the password is not your broadvoice website password but the one you need to get from broadvoice support. Randy Greg Hill wrote: On Tue, 15 Feb 2005, Max Clark wrote: I have experimented with several configs based on different pages and threads but nothing is working. How do I properly configure my broadvoice account? [general] register = [EMAIL PROTECTED]:pass:[EMAIL PROTECTED] the register I'm using looks like this: register = 310584:pass@sip.broadvoice.com [broadvoice] type=peer host=sip.broadvoice.com secret=pass fromuser=310584 fromdomain=sip.broadvoice.com context=incoming dtmfmode=inband canreinvite=no nat=yes qualify=yes try: [broadvoice] type=peer username=310584 secret=pass host=sip.broadvoice.com port=5060 context=incoming fromuser=310584 fromdomain=sip.broadvoice.com dtmfmode=inband canreinvite=no insecure=very permit=147.135.8.128/32 qualify=yes and adjust your permit= line to match the IP of the BV proxy you've set in your /etc/hosts (proxy.[chi|lax|dca|one other I forgot].broadvoice.com). Try a blend of this stuff with whatever the most recent recommendation on their support page says. Greg ___ 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 ___ 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] DTMF inband detection improvement
On Feb 16, 2005, at 10:34 AM, Steve Underwood wrote: BTW, Steve, if you're still reading, what is the RADIO_RELAX option intended to be for in dsp.c? It is something someone else added to the code to make the detection criteria in relaxed mode even more relaxed. If setting that helps, something in your channel must be causing some serious filtering of low frequencies. Can you try logging the audio to a file, and send it to me for analysis? chan_spy, or something like that, should do the job. Actually, it was Florian that posted about this option. I haven't tried it (spent an awful lot of time last week compiling different configurations of stable, head, patches...taking a break this week). This is Florian said: On Feb 15, 2005, at 1:12 PM, Florian Lefeuvre wrote: I find the compilation option RADIO_RELAX. this option change a threshold in DTMF detection (function dtmf_detect in dsp.c) I remark an big improvement in the detection of the dtmf over GSM. have you ever test this option? RADIO is obscur for me, does it mean all wireless device? Florian -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Inter-asterisk conferencing delays - IAX2 configuration problem?
Title: Inter-asterisk conferencing delays - IAX2 configuration problem? Hi We are having a significant ( 1 sec) delay in a multi-asterisk conference, with IAX2 legs connecting meetme on different boxes. All the other legs are PSTN (TE410P). The example configuration Slave box 1 meetme --- IAX2 --- Master box meetme --- IAX2 --- Slave box 2 meetme The delay is between Slave box 1 and Slave box 2 The primary suspect is our iax configuration (changing trunking from 'yes' to 'no' didn't help). Could you, please, point to IAX2 related parameters that can make a difference? [general] port=5036 disallow=all allow=ulaw tos=lowdelay [iaxcontext] type=user context=default_iax deny=0.0.0.0/0.0.0.0 permit=10.50.4.0/255.255.255.0 ;trunk=yes ; Use IAX2 trunking with this host trunk=no ; Do not use IAX2 trunking with this host Thanks. Alex Zarubin ___ 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] Re: X100P problems
it only reloads asterisk. in 0.6 it also reloads FOP. --- David Josephson [EMAIL PROTECTED] wrote: Yes - the problem was a missing signalling line in zapata.conf. Now in and out work. Also, it was news that reload from the console doesn't reflect changes made in zaptel and zapata.conf entries. Any other config files that it doesn't reload? From: [EMAIL PROTECTED] [EMAIL PROTECTED] Did you go into AMP and configure some place for incomming calls to go? --- David Josephson [EMAIL PROTECTED] wrote: Is there a configuration difference for clone X100P cards versus compatible? I have a similar problem to what David Shaw posted earlier ___ 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 __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ 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] IAX Hardphone AT-320EE
AT-320EE Anyone try these? Do they work? any reviews? I couldn't find jack on google.. ___ 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] IAX Hardphone AT-320EE
I was also looking at these this morning but couldnt find any info. I am interested in an IAX hardphone that works. Matt Schulte wrote: AT-320EE Anyone try these? Do they work? any reviews? I couldn't find jack on google.. ___ 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] Sphinx
In a production environment, I would not attempt to run Sphinx on the same computer as Asterisk A few users interacting with Sphinx could consume all of the server's resources and then some. Same goes for DMBS servers, One big N-way join could tie up a CPU for tens of seconds. --- Mark Kidd [EMAIL PROTECTED] wrote: Has anybody managed to implement Sphinx in their * system reasonably painlessly. if so: does it cause any problems with normal * operations. does it place any sort of constant heavy load on the machine. are there options for simple vs advanced implementations. all i am looking for is basicaly for a person to say a branch name. ie: johannesburg thanks ___ 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? All your favorites on one personal page Try My Yahoo! http://my.yahoo.com ___ 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] Solaris 10
Wang Xiangzhou wrote: Sun claims that Linux apps can run on Solaris 10 natively. Is there anyone to run Asterisk on Solaris 10 and what the results are. Thanks, William why not just compile asterisk on sol10? Ming-Wei ___ 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] Asterisk@Home 0.6 Released
New features include Festival text to speech and a new Web Conferencing GUI. There are also numerous small fixes and enhancements. http://asteriskathome.sourceforge.net/ __ Do you Yahoo!? All your favorites on one personal page Try My Yahoo! http://my.yahoo.com ___ 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] WLAN-Voip phones anyone?
Hello, Does anyone here use any WLAN phones with asterisk? Are there any posts about problems, security (and prices in germany)? Bye, Olaf -- Olaf Klein Adimus Beratungsges. für System- und Netzwerkadministration mbH Harpener Hellweg 41 44805 Bochum Tel. 0234-95015-13 Fax. 0234-95015-29 Mobil 0177-3264501 ___ 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] speech recognition V 2.0
Sphinx and Festival are good projects. The last I worked with sphinx I was told that it would need modifications to make it more grammar aware, but that was 2 years ago and things may have improved. If not then Sphinx people please let me know when you will add grammars natively or refer me a grammar based engine. Sphinx and Festival are in fact the current state of the art. you are not likely to find anything better. Sphinx can return a probibility network. You can then attempt to parse paths through the network and use the first path (searching in probibillity order) that parses correctly. You can use a LEX/YACC parser and do well enough. (Get the O'Reilly LEX/YACC book. It's easy to use.) I'm impressed with YACC's performance. I have an application with hundres of grammar rules that runs as fast as UNIX's wc utility. Users _can_ learn the subset of grammer. Remember the game zork or the other text based adventure games? People caught on to the limited subset of English. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ 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] Solaris 10
Yeah, I've been running asterisk 1.0.3 and 1.0-RC1 before that on Solaris 10. I'm only using it for personal use though. Really I'm just using SIP to a sipura, broadvoice and freeworlddialup with voice mail and such. It works fine for my purposes but I can't attest to testing it well enough for someone to use in a production environment. - logan On Wed, Feb 16, 2005 at 08:06:27PM +0100, Ming-Wei Shih wrote: Wang Xiangzhou wrote: Sun claims that Linux apps can run on Solaris 10 natively. Is there anyone to run Asterisk on Solaris 10 and what the results are. Thanks, William why not just compile asterisk on sol10? Ming-Wei ___ 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] Queue strategy
As for a good way to log him out, you can set autologout=20 in agents.conf in order to logout agents whose phone rings more than 20 seconds. Ideally, this should be set to the same value as the timeout on the queue that the agent is not answering. As for emailing their manager - Not a built-in way, but you could check out some real-time monitoring tools like the flash operator panel - http://www.voip-info.org/wiki-Asterisk+Flash+Operator+Panel I like the way that queues and agents are displayed there, and it really would give call center managers a great deal of information at-a-glance about their agents' activities. Regards, Todd ___ 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] Using Hylafax and Digium T100P
You could always run another T100P into a HylaFAX-run T1 fax modem. That way you can use your T1 for faxing. Could you explain a litter further? Thanks. On 2005.02.16 09:00 Brian M. Arlinghaus wrote: My problem is that the 23 channels are going into asterisk. It seems that there is no way to pick off a couple of them and use them for faxes. Basically, I have 23 phone lines and can't use them for faxing. ___ 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] Using Hylafax and Digium T100P
Lee So the drop/insert channel bank will pick off a few of the channels and send the rest to asterisk? Is this some sort of Adtran product? What about DIDs? Thanks, Brian - Original Message - From: Jon Pounder [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 12:35 PM Subject: Re: [Asterisk-Users] Using Hylafax and Digium T100P On 2005.02.16 09:00 Brian M. Arlinghaus wrote: My problem is that the 23 channels are going into asterisk. It seems that there is no way to pick off a couple of them and use them for faxes. Basically, I have 23 phone lines and can't use them for faxing. You could always run another T100P into a HylaFAX-run T1 fax modem. That way you can use your T1 for faxing. you need a drop and insert channel bank. T1 from telco to channel bank, second T1 from channel bank to asterisk, analog ports on channel bank. by drop and insert you can either connect the analog ports to the telco t1 or the asterisk t1 and the rest of the channels pass through between the two t1's Lee. ___ 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 !DSPAM:42137ed031451817322028! Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ 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
[Asterisk-Users] Polycom MGCP firmware
I have a Polycom 400 with H.323 firmware. I know it is not capable of loading the SIP firmware. Anybody know if I can get the MGCP firmware (and maybe the bootloader) from somewhere? ___ 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: Why Asterisk can't cope with silence suppression?
Essentially its because * has been architected to send an rtp packet after receiving a packet. If * never see's and incoming rtp packet, then it won't send an rtp packet (which usually contains some amount of audio). Thus choppy audio in one direction. So why cant * just play comfort noise when it doesnt see any rtp packets in a particular bearer channel? Unless I am missing something fundamental this doesnt seem to be a huge architectural change. Id have to agree that a lack of proper vad support is a major shortcoming. ___ 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] Help Please!!!!
I have a asterisk server with 6 Cisco ATA connected in SIP. My problem is that one of them is dropping calls an I can't figure out what is the problem; I had made a SIP DEBUG PEER 1088 that is the peer with the problem. Any help will be appreciate Thanks Erick Weber VoIP*CLI sip debug peer 1088 SIP Debugging Enabled for IP: 201.133.170.82:5060 Peer RTP is at port 192.168.1.69:0 Peer RTP is at port 192.168.1.69:0 -- Executing Dial(SIP/404-cbc9, SIP/1088|60|tr) in new stack We're at XXX.XXX.XXX.XXX port 17506 Answering/Requesting with root capability 256 12 headers, 8 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:43:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 1679 1679 IN IP4 XXX.XXX.XXX.XXX s=session c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 17506 RTP/AVP 18 a=rtpmap:18 G729/8000 a=silenceSupp:off - - - - (NAT) to 201.133.170.82:5060 -- Called 1088 -- SIP/1088-ec82 is ringing Found RTP audio format 18 Found RTP audio format 101 Peer RTP is at port 192.168.1.2:0 Found description format G729 Found description format telephone-event Capabilities: us - 0x100(G729A), peer - audio=0x100(G729A)/video=0x0(EMPTY), combined - 0x100(G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp set_destination: Parsing sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp for address/port to send to set_destination: set destination to 192.168.1.2, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED];tag=939809556 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 201.133.170.82:5060 -- SIP/1088-ec82 answered SIP/404-cbc9 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 Using latest request as basis request Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0711b1d6 Content-Length: 0 to 201.133.170.82:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Using latest request as basis request Sending to 192.168.1.2 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: sip:[EMAIL PROTECTED];expires=120 Date: Wed, 16 Feb 2005 00:43:46 GMT Content-Length: 0 to 201.133.170.82:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:201.133.170.82 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK0972cae7 From: asterisk sip:[EMAIL PROTECTED];tag=as59adf4c2 To: sip:201.133.170.82 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:43:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 201.133.170.82:5060 Destroying call '[EMAIL PROTECTED]' set_destination: Parsing sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp for address/port to send to set_destination: set destination to 192.168.1.2, port 5060
Re: [Asterisk-Users] Using Hylafax and Digium T100P
On 2005.02.16 11:20 Brian M. Arlinghaus wrote: You could always run another T100P into a HylaFAX-run T1 fax modem. That way you can use your T1 for faxing. Could you explain a litter further? Thanks. Well, you can do something like this: T1 -- TE405P(1) -- Asterisk -- TE405P(2) -- Patton 2977 -- HylaFAX So you've got a TE405P with 4 ports on it. Use one port to receive the T1 from the telco. Use another port to connect the T1 fax modem (Patton 2977 in this case). When Asterisk gets a call on a fax DID, then it just Dial()s it through to the other port, handing the call off to the fax modem. This way Asterisk shouldn't be involved in anything other than forwarding the digital call packets to the fax device. So, if you don't have another port, then you'd need to get an additional T100P to link up the T1 fax modem with Asterisk. Lee. ___ 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] Agent Logoff not generating event messages
CVS Head 02/02/2005 from the CLI command line, the command Agent logoff Agent/agentnum soft does log the agent out, but does not generate any manager events. The AgentLogoff and AgentCallbacklogin apps do generate such events. Should the command line agent logoff also generate a manager event ? Julian. ___ 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] Sip Notify PAP2-NA?
I am using mysql sipfriends and can't seem to get the MWI to work. From what I've read it seems this is not supported with that dynamic system, and probably never will be. I was thinking of just setting a cron job or something to check every minute for voicemail and set our sip NOTIFY messages as needed. Also, the PAP2-NA has the ability to reboot via a sip notify and I would like to be able to do that. I have seen something to do this on some soft phones, but have not been able to get it to work on the PAP2. Anyone have any experience getting MWI to work with dynamic sipfriends or sending custom sip notify messages to PAP2's? ___ 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] zaphfc buffer underflow/overflow messages
I get a ton of these messages, a pair every 4 or 5 mins. Is it a problem? I am wondering where they come from and if they are important. I have a zaphfc card running in TE mode connected to a PBX. Feb 16 20:23:04 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:23:04 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 49, 49 Feb 16 20:24:09 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:24:09 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 46, 46 Feb 16 20:25:13 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:25:13 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 51, 51 Feb 16 20:26:18 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:26:18 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 48, 48 Feb 16 20:27:23 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:27:23 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 45, 45 Feb 16 20:28:27 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:28:27 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 50, 50 Feb 16 20:29:32 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:29:32 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 47, 47 Feb 16 20:30:36 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:30:36 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 52, 52 Feb 16 20:31:41 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:31:41 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 49, 49 ___ 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] When callerid changes its value ?
Hi, I'm reading a lot of stuff about callerid problems, but couldn't find any logical explanation of Asterisk behaviour with callerid. When I receive incoming call, caller info seems ok, but when transferred to local extension via some macros, callerid gets to 'asterisk'. Does anyone know why and when this happens ? Does equipment like Grandstream behave normally and displays only callerdinum or should caller id name be same as number ? How do you deal easily with caller id if you have several different sip/iax clients ? Thanks in advance, regards, Rob. ___ 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] capiECT problem
Hi,
I'm trying to get capiECT working. I'd like to transfer call to another ISDN
router connected extension and free channel from router to Asterisk.
I get incoming call on CAPI and would liek to transfer it to dialed local
extension - 400 in this case:
[outbound-capi-local]
exten = _4XX,1,NoOp(Transferring to local PBX ISDN number ${EXTEN} on msn
CAPI/${CALLERIDNUM})
exten = _4XX,2,capiHOLD
exten = _4XX,3,capiECT,${CALLERIDNUM:1}:${EXTEN}
When I dial 400, another extension rings, shows right callerid (1st argument
to capiECT), but incoming call gets constant sound and obviously loses
connection. But capi channel is freed. When I lift handset of 400 extension,
asterisk s starts to anounce number that was sent as callerid ...
Any help, hint or working example for capiECT ?
Thanks in advance,
regards,
Rob.
Follows console session:
-- Executing NoOp(CAPI[contr1/7104370]/3, Transferring to local PBX
ISDN number 400 on msn CAPI/0037103780) in new stack
-- Executing capiHOLD(CAPI[contr1/7104370]/3, ) in new stack
-- Executing capiECT(CAPI[contr1/7104370]/3, 037103780:400) in new
stack
-- creating pipe for PLCI=-1
sent CONNECT_REQ MN =0x658
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/3' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/3' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/8' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Executing Hangup(CAPI[contr1/7104370]/3, ) in new stack
== Spawn extension (aa_1, h, 1) exited non-zero on
'CAPI[contr1/7104370]/3'
-- CAPI Hangingup
sent DISCONNECT_REQ PLCI=0x201
-- removed pipe for PLCI = 0x201
___
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[Asterisk-Users] Re: Sangoma A102 cards testing FIXED
+++ Robert Augustyn [15/02/05 15:04 -0500]: May I ask what you did? robert I'm sorry if it appears like Quoted text thats cause i cut pasted it rom my mail to sangoma. but what i did is right there in the mail below Its fixed and working great. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vikram Rangnekar Sent: Tuesday, February 15, 2005 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Sangoma A102 cards testing FIXED Its fixed and working great. I was working on the dma issue that nenad pointed out and when i tried to hdparm -d 1 /dev/hda my harddisk i got a permissioned denied error (I was root) So i started researching a bit more and realised that since i have a sata hdd and running it in IDE mode i cant start dma so i just recompiled the kernel with sata drivers and scsi activated then i change the bio SATA setting to SATA ENHANCED (nothing else seems to work) and changed the root=/dev/hda2 to root=/dev/sda2 in the kernel boot options and i was on my way. now when i tierd hdparm -d 1 /dev/sda i was told that scsi dosent have dma so it didnt matter. Next i started the wanpipe drivers and started asterisk i didnt get any errors so to test th config i used exten = 111,1,Dial(Zap/g1/301) exten = 111,2,Hangup and dialed that extension 111 and the extension 301 rang. i had started pri intense debugging and used show channels to make sure the call went over the e1 channels. i'm still testing it, it seems to be working great right now only error i got was a FCS BAD or somthing like that once. My motherboard is a SuperMicro P4SCI just for your information. regards Vikram (http://www.vicramresearch.com) ___ 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
