Re: [Asterisk-Users] SER redirect
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Can you specify a bit more what you whant to have help with! Best regards jan Sharon wrote: hello, can someone help me with ser redirect to asterisk. any help appreciated. Thanks, AA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - -- +--- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05 ! E-mail: [EMAIL PROTECTED] ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFD3JwXQEpdoflEoIsRAscBAJ9TUT+glfYs7AsIc9VuqsnmeH9SNgCfePY7 BYdhQejNlF1f1p7yKyMAHP8= =Hqh4 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
Vic a écrit : Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. What exactly do you do with these calls? We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. If you are just handling SIP signaling and routing a solution such as SIP Express Router is much more appropriate than Asterisk for this kind of volume. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. It depends on what you are trying to do. If you are transcoding 5,000 simultaneous calls, it's going to cost a lot of money, wether you use Asterisk or not... -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] changing displayed call info on snom 360
Several of my SIP users are in the habit of diverting all their calls to an assistant when they're out of the office. When these calls ring on the assistant's phone, she wants to be able to tell which number they've been forwarded from so that she can say Joe Blow's phone or whatever when she picks up the call. The assistant's phone is a snom 360, which normally just displays the number of the calling party while it's ringing. Snom's FAQ page at http://www.snom.com/wiki/index.php/FAQs suggests that I can send a SIP INFO message to the phone to change the displayed call information. I did a few experiments with a hacked chan_sip.c, but wasn't able to produce any visible effect on the phone. Does anybody have any experience making this snom feature work with Asterisk, or know of any other way to influence the information that's displayed on the phone? Thanks Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
Sure enough we lost ALL sip-sip audio on 1-25 Pulled my hair out for hours before looking here or at the website to find this problem reported... Very greatful to find this I have upgraded to 1.2.3 but still have no sip-sip audio! what?! Now I'm back to contnued hair pulling what culd I possible be missing? Have started over re-compiled and reinstalled rebooted, tried it all over again -- yet cannot get any audio between sip phones on the LAN! I must be doing something stupid but what???! Any pointers appreciated! Thanks! Steve -- Vontage*CLI show version Asterisk 1.2.3 built by root @ Vontage on a i686 running Linux on 2006-01-29 10:38:54 UTC Vontage*CLI show version files File Revision func_callerid.c Revision: 7221 cdr_custom.c Revision: 7221 cdr_manager.c Revision: 7221 cdr_csv.c Revision: 7221 format_g723.c Revision: 7221 format_jpeg.c Revision: 7221 format_au.c Revision: 7221 format_sln.c Revision: 7221 format_ilbc.c Revision: 7221 format_g726.c Revision: 7221 format_h263.c Revision: 7221 format_pcm_alaw.c Revision: 7819 format_g729.c Revision: 7221 format_pcm.c Revision: 7819 format_vox.c Revision: 7221 format_wav_gsm.c Revision: 7221 format_wav.c Revision: 7221 format_gsm.c Revision: 7221 codec_g726.c Revision: 7221 codec_a_mu.c Revision: 7221 codec_alaw.c Revision: 7221 codec_ulaw.c Revision: 7221 codec_adpcm.c Revision: 7221 codec_lpc10.c Revision: 7221 codec_gsm.c Revision: 7221 codec_ilbc.c Revision: 7221 app_sms.c Revision: 7634 app_page.cRevision: 7274 app_zapscan.c Revision: 7221 app_zapbarge.cRevision: 7221 app_flash.c Revision: 7221 app_meetme.c Revision: 8194 app_zapras.c Revision: 7221 app_mixmonitor.c Revision: 7740 app_directed_pickup.c Revision: 7550 app_externalivr.c Revision: 7634 app_dictate.c Revision: 7221 app_settransfercapability Revision: 7221 app_chanspy.c Revision: 7740 app_readfile.cRevision: 7221 app_md5.c Revision: 7221 app_setrdnis.cRevision: 7221 app_while.c Revision: 7221 app_waitforsilence.c Revision: 7605 app_dumpchan.cRevision: 7221 app_realtime.cRevision: 7221 app_math.cRevision: 7221 app_forkcdr.c Revision: 7221 app_test.cRevision: 7221 app_verbose.c Revision: 7221 app_userevent.c Revision: 7221 app_alarmreceiver.c Revision: 7221 app_talkdetect.c Revision: 7221 app_controlplayback.c Revision: 7221 app_txtcidname.c Revision: 7221 app_groupcount.c Revision: 7221 app_exec.cRevision: 7221 app_sendtext.cRevision: 7221 app_nbscat.c Revision: 7221 pp_eval.cRevision: 7221 app_ices.cRevision: 7221 app_random.c Revision: 7221 app_setcdruserfield.c Revision: 7221 app_read.cRevision: 7221 app_cut.c Revision: 7497 app_sayunixtime.c Revision: 7221 app_hasnewvoicemail.c Revision: 7608 app_cdr.c Revision: 7221 app_setcidnum.c Revision: 7221 app_transfer.cRevision: 7221 app_enumlookup.c Revision: 7221 app_chanisavail.c Revision: 7221 app_db.c Revision: 7221 app_privacy.c Revision: 7771 app_waitforring.c Revision: 7221 app_lookupblacklist.c Revision: 7221 app_softhangup.c Revision: 7221 app_authenticate.cRevision: 7221 app_macro.c Revision: 7221 app_lookupcidname.c Revision: 7221 app_setcidname.c Revision: 7221 app_parkandannounce.c Revision: 7221 app_senddtmf.cRevision: 7221 app_queue.c Revision: 8445 app_festival.cRevision: 8140 app_setcallerid.c Revision: 7221 app_zapateller.c Revision: 7221 app_milliwatt.c Revision: 8232 app_getcpeid.cRevision: 7221 app_adsiprog.cRevision: 7221 app_disa.cRevision: 7221 app_url.c Revision: 7221 app_image.c Revision: 7221 app_record.c Revision: 7221 app_echo.cRevision: 7221 app_system.c Revision: 7221 app_mp3.c Revision: 7221 app_directory.c Revision: 7221 app_voicemail.c Revision: 7999 app_playback.cRevision: 7221 app_dial.cRevision: 8608 chan_zap.cRevision: 8573 chan_phone.c
Re: [Asterisk-Users] SER redirect
Jan Saell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Can you specify a bit more what you whant to have help with! I guess it is the usual question nobody wants to answer, right? (Internet) == port 5060 = SER redirect EVERYTHING to port 5062 = Asterisk bye Ronald Wiplinger Best regards jan Sharon wrote: hello, can someone help me with ser redirect to asterisk. any help appreciated. Thanks, AA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - -- +--- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05 ! E-mail: [EMAIL PROTECTED] ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFD3JwXQEpdoflEoIsRAscBAJ9TUT+glfYs7AsIc9VuqsnmeH9SNgCfePY7 BYdhQejNlF1f1p7yKyMAHP8= =Hqh4 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: (Un)PauseQeueMamber usage
On 1/28/06, Kevin Bockman [EMAIL PROTECTED] wrote: Joe wrote: Thanks for the reply BJ. Your example makes sense for out-bound traffic, but what about calls transferred from a queue to an agent? When an agent receives a call, they will be marked busy anyways as long as you are using agent members for the queue. (member = Agent/1000) That's only true for the 1 queue for which the Agent received a call when using callback mode. If the Agent is a member of another queue and they are next to receive a call, unless they are paused, they will receive that call when using Agents in callback mode. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: (Un)PauseQeueMamber usage
On 1/28/06, Joe [EMAIL PROTECTED] wrote:
Thanks for the reply BJ. Your example makes sense for out-bound traffic, but
what about calls transferred from a queue to an agent?
I plan on setting up agent extensions (if possible via macro) something like
this for example:
exten = 1234,1,PauseQueueMember (|Agent/101)
exten = 1234,2,Dial(Agent/101,tg)
exten = 1234,3,UnPauseQueueMemeber(|Agent/101)
exten = 1234,4,Hangup()
Agents will login using AgentCallBackLogin. In the example above, Agent 101
will login from extension 1234. This would work well if Agent 101 was always
sitting at the phone with extension 1234. This will more than likely not be
the case.
Is this what I need:
exten = 1234,1,PauseQueueMember(|Agent/${AGENTBYCALLERID_${CALLERIDNUM}})
exten = 1234,2,Dial(Agent/${AGENTBYCALLERID_${CALLERIDNUM}},tg)
exten =
1234,3,UnPauseQueueMemeber(|Agent/${AGENTBYCALLERID_${CALLERIDNUM}})
exten = 1234,4,Hangup()
Not sure if this is the proper use of this variable or not.
Joe -
Yes, you're close. You may want to use
Agent/${$[AGENTBYCALLERID_${CALLERID(num)}]}) instead to be compliant
with the new CALLERID dial plan function in 1.2+.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: (Un)PauseQeueMamber usage
On 1/29/06, BJ Weschke [EMAIL PROTECTED] wrote:
On 1/28/06, Joe [EMAIL PROTECTED] wrote:
Thanks for the reply BJ. Your example makes sense for out-bound traffic, but
what about calls transferred from a queue to an agent?
I plan on setting up agent extensions (if possible via macro) something like
this for example:
exten = 1234,1,PauseQueueMember (|Agent/101)
exten = 1234,2,Dial(Agent/101,tg)
exten = 1234,3,UnPauseQueueMemeber(|Agent/101)
exten = 1234,4,Hangup()
Agents will login using AgentCallBackLogin. In the example above, Agent 101
will login from extension 1234. This would work well if Agent 101 was always
sitting at the phone with extension 1234. This will more than likely not be
the case.
Is this what I need:
exten = 1234,1,PauseQueueMember(|Agent/${AGENTBYCALLERID_${CALLERIDNUM}})
exten = 1234,2,Dial(Agent/${AGENTBYCALLERID_${CALLERIDNUM}},tg)
exten =
1234,3,UnPauseQueueMemeber(|Agent/${AGENTBYCALLERID_${CALLERIDNUM}})
exten = 1234,4,Hangup()
Not sure if this is the proper use of this variable or not.
Joe -
Yes, you're close. You may want to use
Agent/${$[AGENTBYCALLERID_${CALLERID(num)}]}) instead to be compliant
with the new CALLERID dial plan function in 1.2+.
Also - use the 'h' extensions instead of putting UnPauseQueueMember
after the Dial here because in some cases, the call will end and
UnPauseQueueMember won't get executed.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
To check if it's the same problem, set your system clock back 2 weeks. If it gets better, then the upgrade didn't take. If it doesn't get better, it's something else. --Rob -Original Message- Very greatful to find this I have upgraded to 1.2.3 but still have no sip-sip audio! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as SIP endpoint ?
Hello, i want to use asterisk as a ZAP-FXO / SIP gateway. It works fine when I use a SIP provider and register my Asterisk as client there - incoming calls are routed to an extension in a specified context. What I want to do now is to not use the SIP provider and make asterisk accept calls directly at sip:[EMAIL PROTECTED]:5060. The best would be associate [EMAIL PROTECTED] with an extension, [EMAIL PROTECTED] with an other extension in the dialplan. I did not figure out from the docs how to configure this scenario. Peter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as SIP endpoint ?
On 11:25, Sun 29 Jan 06, Peter Molnar wrote: Hello, i want to use asterisk as a ZAP-FXO / SIP gateway. It works fine when I use a SIP provider and register my Asterisk as client there - incoming calls are routed to an extension in a specified context. What I want to do now is to not use the SIP provider and make asterisk accept calls directly at sip:[EMAIL PROTECTED]:5060. The best would be associate [EMAIL PROTECTED] with an extension, [EMAIL PROTECTED] with an other extension in the dialplan. I did not figure out from the docs how to configure this scenario. In sip.conf in the [general] part you have: context= In that context you should have something like: exten = foo,1,Dial(SIP/my_phone) exten = bar,1,Dial(SIP/wifes_phone) or if you want to do more stuff you can use this setup to Goto a specific context for foo and bar: exten = foo,1,Goto(foo-incoming,s,1) Have fun -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Real-time: username
I try to set the username to something useful, like Peter, but it remains the value of 621 1. I set username in the record of name = 621 to Peter Pan 2. I search for this record and found it is set: name=621 and username=Peter 3. I go to Asterisk CLI sip show peers and find the record as: *CLI sip show peers Name/username HostDyn Nat ACL Port Status 621/621192.168.250.76 D N 5060 OK (65 ms) 4. I search for this record again and found it is set: name=621 and username=621 Bug or feature? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wifi phone set-up
I got some troubles with my wifi phone. I used to have set it to: Proxy server: Proxy IP: sip.elmit.com Port: 5060 Expire time 1200 Outbound proxy Proxy IP fwdnat.pulver.com Port:: 5082 User account Phone: 610 Username: 610 User Pwd: since a while this set-up does not work anymore!!! I believe that fwdnat.pulver.com does not forward anymore to my server. I wanted to change it to stun and changed it to: Proxy server: Proxy IP: sip.elmit.com Port: 5060 Expire time 1200 Outbound proxy Proxy IP stun01.sipphone.com Port:: 3478 User account Phone: 610 Username: 610 User Pwd: However, that does also not work. Can anybody give me a hint, how it could work? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Moprobe Zaptel error
Hi, i'm trying to install a compatible modem to act as a X100P, and i would appreciate some help here, this is what is hapening when i modprobe zaptel: # modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.11-6mdk-i586-up-1GB/misc/zaptel.ko): Invalid module format firstly, i did: 'make clean' then 'make linux26' and then 'make install' into zaptel directory Also, in the Linux kernel source Makefile, i replaced: EXTRAVERSION = -6mdkcustom with this: EXTRAVERSION = -6mdk-i586-up-1GB See the error in /var/log/messages: Jan 29 10:41:46 maxilinux kernel: zaptel: version magic '2.6.11-6mdk-i586-up-1GB 686 gcc-3.4' should be '2.6.11-6mdk-i586-up-1GB 586 gcc-3.4' Apparently the difference comes from '686' when it should be '586' How do i fix this? (i've been reading but i did not find the answer to this) To give you some info: # uname -r 2.6.11-6mdk-i586-up-1GB # uname -a Linux maxilinux 2.6.11-6mdk-i586-up-1GB #1 Tue Mar 22 15:46:07 CET 2005 i586 AMD-K6(tm) 3D processor unknown GNU/Linux Thanks, Maxi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] file.c:509 ast_openstream_full: File 100 does not exist in any format
Hi all, look at these lines. I created a queue named info when a caller (extension 86) place a call he is put on queue he sould hear MOH . What's the meaning of : Jan 29 14:35:30 WARNING[2591]: file.c:509 ast_openstream_full: File 100 does not exist in any format Jan 29 14:35:30 WARNING[2591]: file.c:821 ast_streamfile: Unable to open 100 (format ulaw): No such file or directory Regards Harry Jan 29 14:34:43 WARNING[2568]: pbx.c:2403 __ast_pbx_run: Timeout, but no rule 't' in context 'info' -- Executing Answer(SIP/86-a9b4, ) in new stack -- Executing Queue(SIP/86-a9b4, info|tn||100) in new stack -- Started music on hold, class 'default', on channel 'SIP/86-a9b4' -- outgoing agentcall, to agent '101', on 'Local/[EMAIL PROTECTED],1' -- Executing Answer(Local/[EMAIL PROTECTED],2, ) in new stack -- Called Agent/101 -- Agent/101 answered SIP/86-a9b4 Jan 29 14:35:30 WARNING[2591]: file.c:509 ast_openstream_full: File 100 does not exist in any format Jan 29 14:35:30 WARNING[2591]: file.c:821 ast_streamfile: Unable to open 100 (format ulaw): No such file or directory -- Stopped music on hold on SIP/86-a9b4 -- Executing Dial(Local/[EMAIL PROTECTED],2, Sip/85|30|t) in new stack -- Called 85 -- SIP/85-7874 is ringing == Spawn extension (support, info, 2) exited non-zero on 'SIP/86-a9b4' == Spawn extension (info, 85, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modprobe Zaptel error
Hi, i'm trying to install a compatible modem to act as a X100P, and i would appreciate some help here, this is what is hapening when i modprobe zaptel: # modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.11-6mdk-i586 -up-1GB/misc/zaptel.ko): Invalid module format firstly, i did: 'make clean' then 'make linux26' and then 'make install' into zaptel directory Also, in the Linux kernel source Makefile, i replaced: EXTRAVERSION = -6mdkcustom with this: EXTRAVERSION = -6mdk-i586-up-1GB See the error in /var/log/messages: Jan 29 10:41:46 maxilinux kernel: zaptel: version magic '2.6.11-6mdk-i586-up-1GB 686 gcc-3.4' should be '2.6.11-6mdk-i586-up-1GB 586 gcc-3.4' Apparently the difference comes from '686' when it should be '586' How do i fix this? (i've been reading but i did not find the answer to this) To give you some info: # uname -r 2.6.11-6mdk-i586-up-1GB # uname -a Linux maxilinux 2.6.11-6mdk-i586-up-1GB #1 Tue Mar 22 15:46:07 CET 2005 i586 AMD-K6(tm) 3D processor unknown GNU/Linux Thanks, Maxi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
On Sun, 2006-01-29 at 17:24 +1000, Rob Thomas wrote: [snip] If you do, honestly, need to handle 5k calls, you’d probably have to have a bank of Cisco 5850s doing the termination Or have a look at the Lucent APX8100 box for some added carrier class humpf. Supports more than 8000 DS0's (channels) and does transcoding in hardware DSP's so well suited to handle your 5000 concurrent calls and you don't need a stack of them like with the Cisco 5850. Weblink: http://www.lucent.com/products/subcategory/0,,CTID+2013-STID+10443-LOCL +1,00.html Datasheet: http://www.lucent.com/livelink/090094038000eb91_Brochure_datasheet.pdf Like Rob I'd love to sell this to you but I doubt Lucent would even pick up the phone to answer my how to become a VAR enquiry. Best contact them directly :) Regards, Patrick (no affiliation with Lucent) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to remove first ring tone on FXO?
Hi everybody, Every time callers reach my FXO port,asterisk producesone ring tone just beforeit executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to remove first ring tone on FXO?
It is waiting for the CalledID information. Set usecallerid=no and that should do it for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto RachmadSent: Sunday, January 29, 2006 9:40 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] How to remove first ring tone on FXO? Hi everybody, Every time callers reach my FXO port,asterisk producesone ring tone just beforeit executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to remove first ring tone on FXO?
Thanks Alexander, Ijusttried that, but itdoesn't help. There is still one ring tone produced before asterisk executes Answer(). And thereis nocaller ID being forwarded to the destination channel, which actually I need. That is why I have usecallerid set to yes. Cheers, Anto - Original Message - From: Alexander Lopez To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 3:54 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? It is waiting for the CalledID information. Set usecallerid=no and that should do it for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto RachmadSent: Sunday, January 29, 2006 9:40 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] How to remove first ring tone on FXO? Hi everybody, Every time callers reach my FXO port,asterisk producesone ring tone just beforeit executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Patrick Sent: Sunday, January 29, 2006 7:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout question On Sun, 2006-01-29 at 17:24 +1000, Rob Thomas wrote: [snip] If you do, honestly, need to handle 5k calls, you'd probably have to have a bank of Cisco 5850s doing the termination Or have a look at the Lucent APX8100 box for some added carrier class humpf. Supports more than 8000 DS0's (channels) and does transcoding in hardware DSP's so well suited to handle your 5000 concurrent calls and you don't need a stack of them like with the Cisco 5850. Weblink: http://www.lucent.com/products/subcategory/0,,CTID+2013-STID+10443-LOCL +1,00.html Datasheet: http://www.lucent.com/livelink/090094038000eb91_Brochure_datasheet.pdf Like Rob I'd love to sell this to you but I doubt Lucent would even pick up the phone to answer my how to become a VAR enquiry. Best contact them directly :) Regards, Patrick (no affiliation with Lucent) The original poster of this message stated in an earlier message that the calls would be handed off to him SIP, so the media conversion is being done buy an upstream carrier, presumably on a Lucent or Sonus. With the growing availability of SIP origination and termination, high density channels banks like the APX8000 are becoming items only needed by wholesale carriers. Of course this varies by geographical region, but to use a APX 8000 you need at least PRI service over DS1/E1, and ideally PRI service over DS3/E3. The challenge I see with a 5000 INBOUND call setup originated SIP is that the calls will need to be load balanced across many * boxes, no 1 asterisk box is going to take 5000 CONCURRENT calls (500 would impress me). I would suggest; Check to see if the SIP origination provider can give you a round robin delivery of calls over 10 or so * boxes (IP addresses), or find an external method of doing it yourself (like a smart session border controller). IF the calls are terminated to hardphones or softphones (as opposed to purely IVR), make sure you can do RTP re-invites so, when appropriate, the RTP stream is offloaded from * (but consider the impact of doing so). Calculate bandwidth needs carefully - 5000 * 70-75kbps (a/ulaw plus packet overhead) requires a GIG-E IP link from you SIP provider and some very robust networking in between. Terminating 5000 calls on * is relatively uncharted ground, there MAY be some others doing it, but good luck getting them to reveal the company jewels. At the very least, this type of implementation would require a team of the VERY BEST asterisk consultants - might want to call Mark himself if you are serious. Damon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to remove first ring tone on FXO?
Anto, Callerid delays answer until after the first ring, I would suggest you are either not subscribing to your telco for caller id or similar. The advice you got was correct. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad Sent: Sunday, 29 January 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to remove first ring tone on FXO? Thanks Alexander, Ijusttried that, but itdoesn't help. There is still one ring tone produced before asterisk executes Answer(). And thereis nocaller ID being forwarded to the destination channel, which actually I need. That is why I have usecallerid set to yes. Cheers, Anto - Original Message - From: Alexander Lopez To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 3:54 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? It is waiting for the CalledID information. Set usecallerid=no and that should do it for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad Sent: Sunday, January 29, 2006 9:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to remove first ring tone on FXO? Hi everybody, Every time callers reach my FXO port,asterisk producesone ring tone just beforeit executes Answer(). How to remove this? I have commented #define RINGBEGIN on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] username not stabled?
vpbx*CLI sip show peers Name/username HostDyn Nat ACL Port Status 621/621192.168.250.76 D N 5060 OK (65 ms) 626/626192.168.250.109 D N 5060 OK (180 ms) 616/Ronald Softphone (Unspecified)D N 0UNKNOWN 615/Ronald office 192.168.250.103 D N 5060 OK (41 ms) 610/Ronald WiSip (Unspecified)D N 0UNKNOWN 609/Grandstream(Unspecified)D N 0UNKNOWN 608/Note-Pen Softphone (Unspecified)D N 0UNKNOWN 606/Office (Unspecified)D N 0UNKNOWN 605/60561.220.121.19D N 5060 OK (9 ms) 602/60261.220.121.19D N 5060 OK (10 ms) 601/601192.168.250.95 D N 5060 OK (201 ms) a few minutes later: vpbx*CLI sip show peers Name/username HostDyn Nat ACL Port Status 621/621192.168.250.76 D N 5060 OK (65 ms) 626/626192.168.250.109 D N 5060 OK (180 ms) 616/Ronald Softphone (Unspecified)D N 0UNKNOWN 615/615192.168.250.103 D N 5060 OK (41 ms) 610/Ronald WiSip (Unspecified)D N 0UNKNOWN 609/Grandstream(Unspecified)D N 0UNKNOWN 608/Note-Pen Softphone (Unspecified)D N 0UNKNOWN 606/Office (Unspecified)D N 0UNKNOWN 605/60561.220.121.19D N 5060 OK (9 ms) 602/60261.220.121.19D N 5060 OK (10 ms) 601/601192.168.250.95 D N 5060 OK (201 ms) 601, 602, 605, 606, 608, 609, 610, 615 and 616 are in sip.conf 621 and 626 are in Real-time sip_buddies 621 and 626 changes username back from name to number (name) in the database, and never shows it in sip show peer 615 changed username Ronald office to 615, although no change in sip.conf Did anybody else experienced that? *CLI show version Asterisk SVN-trunk-r8447M built by root @ vpbx on a x86_64 running Linux on 2006-01-25 15:33:01 UTC bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to remove first ring tone on FXO?
Thanks a lot Dean, I think thereis a way to remove that ring tone and also still have the caller ID from the incoming call. I have been trying to find that on zaptel.c, chan_zap.c and pbx.c, but I could not find that. Could you pleaseletme know which part of the codes handling that? Cheers, Anto - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 4:23 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? Anto, Callerid delays answer until after the first ring, I would suggest you are either not subscribing to your telco for caller id or similar. The advice you got was correct. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto RachmadSent: Sunday, 29 January 2006 10:09 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] How to remove first ring tone on FXO? Thanks Alexander, Ijusttried that, but itdoesn't help. There is still one ring tone produced before asterisk executes Answer(). And thereis nocaller ID being forwarded to the destination channel, which actually I need. That is why I have usecallerid set to yes. Cheers, Anto - Original Message - From: Alexander Lopez To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 3:54 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? It is waiting for the CalledID information. Set usecallerid=no and that should do it for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto RachmadSent: Sunday, January 29, 2006 9:40 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] How to remove first ring tone on FXO? Hi everybody, Every time callers reach my FXO port,asterisk producesone ring tone just beforeit executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to remove first ring tone on FXO?
Hi Anto, I dont know as I use [EMAIL PROTECTED] these days as so much easier. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad Sent: Sunday, 29 January 2006 10:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to remove first ring tone on FXO? Thanks a lot Dean, I think thereis a way to remove that ring tone and also still have the caller ID from the incoming call. I have been trying to find that on zaptel.c, chan_zap.c and pbx.c, but I could not find that. Could you pleaseletme know which part of the codes handling that? Cheers, Anto - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 4:23 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? Anto, Callerid delays answer until after the first ring, I would suggest you are either not subscribing to your telco for caller id or similar. The advice you got was correct. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad Sent: Sunday, 29 January 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to remove first ring tone on FXO? Thanks Alexander, Ijusttried that, but itdoesn't help. There is still one ring tone produced before asterisk executes Answer(). And thereis nocaller ID being forwarded to the destination channel, which actually I need. That is why I have usecallerid set to yes. Cheers, Anto - Original Message - From: Alexander Lopez To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 3:54 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? It is waiting for the CalledID information. Set usecallerid=no and that should do it for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad Sent: Sunday, January 29, 2006 9:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to remove first ring tone on FXO? Hi everybody, Every time callers reach my FXO port,asterisk producesone ring tone just beforeit executes Answer(). How to remove this? I have commented #define RINGBEGIN on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to remove first ring tone on FXO?
You are wrong, there is no way you can remove the ring, since the ring is something that the callers equipment is generating to the caller, and NOT asterisk. The most you will able to accomplish will be to have just one ring before asterisk picks up. By setting usecallerid to no all you are doing is telling asterisk don't wait for callerid, but since you are using POTS, 2 things will always happen that you can't control: 1. At least part of a ring has to be delivered to Asterisks' FXO port, so that Asterisk knows that there is an incoming call, because inband signalling is used, there is no other way for asterisk to know there is an incoming call. 2. The calling party will always hear at least one ring even if asterisk happens to pick up the line - by mistake - before any ring voltage, because the switch that the line is connected to has already sent a ring indicator to the remote switch, and the remote switch has already started playing the ring tone to the caller, similar to what the playtones(ring) does in Asterisk. Even if it happens to be that you were able to get it once to NOT ring to the caller, it was by mistake that the timing worked out that way, it has nothing to do with what you set in Asterisk. If you have a PRI then you can do that, since if you use Answer as the first command, ring indicator is never sent down the line. On 1/29/06, Aryanto Rachmad [EMAIL PROTECTED] wrote: Thanks a lot Dean, I think there is a way to remove that ring tone and also still have the caller ID from the incoming call. I have been trying to find that on zaptel.c, chan_zap.c and pbx.c, but I could not find that. Could you please let me know which part of the codes handling that? Cheers, Anto - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 4:23 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? Anto, Callerid delays answer until after the first ring, I would suggest you are either not subscribing to your telco for caller id or similar. The advice you got was correct. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad Sent: Sunday, 29 January 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to remove first ring tone on FXO? Thanks Alexander, I just tried that, but it doesn't help. There is still one ring tone produced before asterisk executes Answer(). And there is no caller ID being forwarded to the destination channel, which actually I need. That is why I have usecallerid set to yes. Cheers, Anto - Original Message - From: Alexander Lopez To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 3:54 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? It is waiting for the CalledID information. Set usecallerid=no and that should do it for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad Sent: Sunday, January 29, 2006 9:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to remove first ring tone on FXO? Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented #define RINGBEGIN on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Moprobe Zaptel error
On Sun, Jan 29, 2006 at 01:46:32PM +, Maxi Belino wrote: Hi, i'm trying to install a compatible modem to act as a X100P, and i would appreciate some help here, this is what is hapening when i modprobe zaptel: # modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.11-6mdk-i586-up-1GB/misc/zaptel.ko): Invalid module format firstly, i did: 'make clean' then 'make linux26' and then 'make install' into zaptel directory Also, in the Linux kernel source Makefile, i replaced: EXTRAVERSION = -6mdkcustom with this: EXTRAVERSION = -6mdk-i586-up-1GB See the error in /var/log/messages: Jan 29 10:41:46 maxilinux kernel: zaptel: version magic ' 2.6.11-6mdk-i586-up-1GB 686 gcc-3.4' should be '2.6.11-6mdk-i586-up-1GB 586 gcc-3.4' Apparently the difference comes from '686' when it should be '586' How do i fix this? (i've been reading but i did not find the answer to this) To give you some info: # uname -r 2.6.11-6mdk-i586-up-1GB # uname -a Linux maxilinux 2.6.11-6mdk-i586-up-1GB #1 Tue Mar 22 15:46:07 CET 2005 i586 AMD-K6(tm) 3D processor unknown GNU/Linux I don't know how current Mandriva kernel packages are layed, but you should be able to find the .config file of your existing kernel in that package. Maybe under /usr/share/doc , or maybe in /boot/config* . -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to remove first ring tone on FXO?
Hello CF, I thought that asterisk generated that first ring tone. I didn't think further, especially about what the caller's switching centre is doing when it gets an instruction to reach my number. You are obviously right. That switch will notify the caller (alerting) as soon as it gets a connection confirmed message from my switching centre. And I definitely can not avoid that. Thanks a lot for explaining this. Cheers, Anto - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 29, 2006 5:32 PM Subject: Re: [Asterisk-Users] How to remove first ring tone on FXO? You are wrong, there is no way you can remove the ring, since the ring is something that the callers equipment is generating to the caller, and NOT asterisk. The most you will able to accomplish will be to have just one ring before asterisk picks up. By setting usecallerid to no all you are doing is telling asterisk don't wait for callerid, but since you are using POTS, 2 things will always happen that you can't control: 1. At least part of a ring has to be delivered to Asterisks' FXO port, so that Asterisk knows that there is an incoming call, because inband signalling is used, there is no other way for asterisk to know there is an incoming call. 2. The calling party will always hear at least one ring even if asterisk happens to pick up the line - by mistake - before any ring voltage, because the switch that the line is connected to has already sent a ring indicator to the remote switch, and the remote switch has already started playing the ring tone to the caller, similar to what the playtones(ring) does in Asterisk. Even if it happens to be that you were able to get it once to NOT ring to the caller, it was by mistake that the timing worked out that way, it has nothing to do with what you set in Asterisk. If you have a PRI then you can do that, since if you use Answer as the first command, ring indicator is never sent down the line. On 1/29/06, Aryanto Rachmad [EMAIL PROTECTED] wrote: Thanks a lot Dean, I think there is a way to remove that ring tone and also still have the caller ID from the incoming call. I have been trying to find that on zaptel.c, chan_zap.c and pbx.c, but I could not find that. Could you please let me know which part of the codes handling that? Cheers, Anto - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 4:23 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? Anto, Callerid delays answer until after the first ring, I would suggest you are either not subscribing to your telco for caller id or similar. The advice you got was correct. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad Sent: Sunday, 29 January 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to remove first ring tone on FXO? Thanks Alexander, I just tried that, but it doesn't help. There is still one ring tone produced before asterisk executes Answer(). And there is no caller ID being forwarded to the destination channel, which actually I need. That is why I have usecallerid set to yes. Cheers, Anto - Original Message - From: Alexander Lopez To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 3:54 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? It is waiting for the CalledID information. Set usecallerid=no and that should do it for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad Sent: Sunday, January 29, 2006 9:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to remove first ring tone on FXO? Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented #define RINGBEGIN on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users
[Asterisk-Users] strange performance issue
hi
i just setup a test with asterisk 1.2 to see how many concurrent
calls it could handle, and I came across something quite strange;
with ~1000 calls between two asterisk servers, generated with
[looptest]
exten = _X.,1,GotoIf($[ ${EXTEN} 1000 ]?pickup:dial)
exten = _X.,n(pickup),Answer
exten = _X.,n,Echo
exten = _X.,n(dial),Dial(SIP/looptest/$[ ${EXTEN} + 1])
...ping-pong dialing beween two servers. On one of the servers,
system- and user load was low and fine, but on the other, system load
was around 50%(!). After a little headscratching and debugging, i
found this from oprofile: http://pastebin.ca/39007, and mr Underwood
asked me if I was seeing excessive network traffic. The box has two
NICs, and one of them is connected to a 'mirror' port on the cisco
switch, spewing all traffic from all ports into there. So, yes, I
keep seeing lots of traffic. But then, problem still persists after
'ifconfig eth1 down'. I yet haven't had the possibility to unplug the
cable, as the server is at the server farm, but I'll try tomorrow.
Anyway, given that the problem is directly related to the traffic,
does anyone know how i can avoid this problem _without_ just
unplugging the other NICs cable?
Best regards
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
---
In space, loud sounds, like explosions, are even louder because there
is no air to get in the way.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: (Un)PauseQeueMamber usage
BJ Weschke wrote: On 1/28/06, Kevin Bockman [EMAIL PROTECTED] wrote: Joe wrote: Thanks for the reply BJ. Your example makes sense for out-bound traffic, but what about calls transferred from a queue to an agent? When an agent receives a call, they will be marked busy anyways as long as you are using agent members for the queue. (member = Agent/1000) That's only true for the 1 queue for which the Agent received a call when using callback mode. If the Agent is a member of another queue and they are next to receive a call, unless they are paused, they will receive that call when using Agents in callback mode. BJ, I tested using the following: [testtype1] strategy = leastrecent timeout = 20 autofill=yes member = Agent/11000 member = Agent/11001,2 [testtype2] strategy = leastrecent timeout = 25 autofill=yes member = Agent/11000 member = Agent/11001,2 I login using AgentCallbackLogin with agent 11001. I call into queue testtype1, I get the call. I keep that call on the line and call into queue testtype2, the caller sits in the queue. The agent doesn't get the call until the first call is off of the line. testtype1has 0 calls (max unlimited) in 'leastrecent' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: CLI Agent/11000 (Unavailable) has taken no calls yet Agent/11001 with penalty 2 (Busy) has taken no calls yet No CallersLI testtype2has 1 calls (max unlimited) in 'leastrecent' strategy (0s holdtime), W:1, C:0, A:0, SL:0.0% within 0s Members: CLI Agent/11000 (Unavailable) has taken no calls yet Agent/11001 with penalty 2 (Busy) has taken no calls yet Callers: 1. IAX2/kevin-1 (wait: 0:48, prio: 0) I am using 1.2 release branch rev 8632. Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] simulating a few thousand SIP clients?
hi i'm setting up a rig to handle quite a few SIP clients, so i need a way to simulate, say, 20k SIP ATAs. Does anyone know how? This should of course be as close as possible to 'reality', meaning n% calls per client and the usual REGISTER/OPTION traffic. thanks Best regards Roy Sigurd Karlsbakk [EMAIL PROTECTED] --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
To handle 5000 calls coming in over a PRI, youd need 210 or so T1s or 170 E1s.All of those would generate 320Mega BYTES of data per second (eg, 32Gigabit/sec)[Wai Wu] He not talking about PRI here, but rather SIP to SIP There is no way possible that youre going to pump that amount of data through a PC. Dont care about codecs and dialplans, PCs just dont have that sort of internal bandwidth from peripherals. If all the endpints support reinvite and he is not doing any voice processing at all, there is hardly any data going through the PC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] simulating a few thousand SIP clients?
Set up another * and use the manager api to make lots of calls to the other one. You can even make hundresd calls at a time. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Roy Sigurd Karlsbakk Sent: Sunday, January 29, 2006 1:19 PM To: Asterisk Non-Commercial Discu Subject: [Asterisk-Users] simulating a few thousand SIP clients? hi i'm setting up a rig to handle quite a few SIP clients, so i need a way to simulate, say, 20k SIP ATAs. Does anyone know how? This should of course be as close as possible to 'reality', meaning n% calls per client and the usual REGISTER/OPTION traffic. thanks Best regards Roy Sigurd Karlsbakk [EMAIL PROTECTED] --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
I took a look at the asterisk-1.2.3 Makefile, seems to me that the WARNING is just a list of all the .so files found in the modules directory that aren't also found in a subdirectory, it isn't checking that they were built with the current version. So it's going to complain about the modules that come from asterisk-addons every time make install is run in asterisk, no matter what. Not a big problem once you learn to ignore the message, but people are probably going to keep asking what it means. Julian Lyndon-Smith wrote: Warren, You may only use cdr_addon_mysql.so, but I believe that * normally automatically loads all modules it finds (see modules.conf for autoload=yes). The following modules were found in your modules directory, and 1.2.3 of * did not like them, because you got a warning after compile. In the case of app_rxfax.so and app_txfax.so these must of been compiled with a previous version of *, otherwise it would not have complained about them (I know this, because I had a similar issue). If you have kept the previous version of *, check your makefile for app_txfax and app_rxfax, make the same mods to your 1.2.3 makefile and recompile. * will then not complain about the *fax* modules. You may also need to recompile the asterisk-addons, simply because header files and or libraries may have changed in the core asterisk files. I guess what I am saying is that 1.2.3 of * may work with 1.2.1 of asterisk-addons (that is the latest version as you say), but asterisk-addons would need recompiling as well. If you make cleam;make and make install the asterisk-addons, do you get the same error when you make install asterisk ? Julian. app_addon_sql_mysql.so app_rxfax.so app_saycountpl.so app_striplsd.so app_substring.so app_txfax.so cdr_addon_mysql.so chan_modem_aopen.so chan_modem_bestdata.so chan_modem_i4l.so chan_modem.so format_mp3.so res_config_mysql.so Warren Burstein wrote: Julian Lyndon-Smith wrote: These modules are not part of the standard 1.2.3 release - did you also install the 1.2.3 release of the asterisk-addons package ? The lastest asterisk-addons I found at http://ftp.digium.com/pub/asterisk/ is 1.2.1. The only module I use is cdr_addon_mysql.so. I've been using it with 1.2.2 and 1.2.3 without any problem other than the message during make install, which I just ignore. Is there a need for an update to asterisk-addons? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to get IP of eth0
Hi all, Im trying to set up my asterisk server, but Im having a few problems. My server is running with a public IP address. When I want to set up a call with a softphone in my private network behind a router Im always having an error message. In the CLI session we get a message when the softphone starts up. But after that we get immediately the message Unable to get IP of eth0 : cannot assign requested addresses. Im using NAT so I changed in my router the port forwarding. So the SIP/RTP ports are forwarding to my PC and theres just one client in my network. Can somebody help me? Thanks, Sofie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HandyTone 488 ata?
Anyone tried to muck around with using the 488 for both fxs and fxo with asterisk? I've been playing with one for the last couple of days, and it looks like its a little lower quality then the spa3k. No gain settings, echo canceller is less then ideal on long analog pstn loops, etc. Anyone with good experiences? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] simulating a few thousand SIP clients?
sure, but I need to simulate the SIP REGISTER and OPTION traffic sent by ATAs as well. Best regards Roy Sigurd Karlsbakk [EMAIL PROTECTED] --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. On Jan 29, 2006, at 7:26 PM, Wai Wu wrote: Set up another * and use the manager api to make lots of calls to the other one. You can even make hundresd calls at a time. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Roy Sigurd Karlsbakk Sent: Sunday, January 29, 2006 1:19 PM To: Asterisk Non-Commercial Discu Subject: [Asterisk-Users] simulating a few thousand SIP clients? hi i'm setting up a rig to handle quite a few SIP clients, so i need a way to simulate, say, 20k SIP ATAs. Does anyone know how? This should of course be as close as possible to 'reality', meaning n% calls per client and the usual REGISTER/OPTION traffic. thanks Best regards Roy Sigurd Karlsbakk [EMAIL PROTECTED] --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Moprobe Zaptel error
yes, i do have /boot/config so then? what should i do? thanks again. Maxi2006/1/29, Tzafrir Cohen [EMAIL PROTECTED]: On Sun, Jan 29, 2006 at 01:46:32PM +, Maxi Belino wrote: Hi, i'm trying to install a compatible modem to act as a X100P, and i would appreciate some help here, this is what is hapening when i modprobe zaptel: # modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.11-6mdk-i586-up-1GB/misc/zaptel.ko): Invalid module format firstly, i did: 'make clean' then 'make linux26' and then 'make install' into zaptel directory Also, in the Linux kernel source Makefile, i replaced: EXTRAVERSION = -6mdkcustom with this: EXTRAVERSION = -6mdk-i586-up-1GB See the error in /var/log/messages: Jan 29 10:41:46 maxilinux kernel: zaptel: version magic ' 2.6.11-6mdk-i586-up-1GB 686 gcc-3.4' should be '2.6.11-6mdk-i586-up-1GB 586 gcc-3.4' Apparently the difference comes from '686' when it should be '586' How do i fix this? (i've been reading but i did not find the answer to this) To give you some info: # uname -r 2.6.11-6mdk-i586-up-1GB # uname -a Linux maxilinux 2.6.11-6mdk-i586-up-1GB #1 Tue Mar 22 15:46:07 CET 2005 i586 AMD-K6(tm) 3D processor unknown GNU/LinuxI don't know how current Mandriva kernel packages are layed, but youshould be able to find the .config file of your existing kernel in that package. Maybe under /usr/share/doc , or maybe in /boot/config* .--Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's[EMAIL PROTECTED] | |bestICQ# 16849755 | | friend___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Thanks! Steve To check if it's the same problem, set your system clock back 2 weeks. If it gets better, then the upgrade didn't take. If it doesn't get better, it's something else. --Rob -Original Message- Very greatful to find this I have upgraded to 1.2.3 but still have no sip-sip audio! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to remove first ring tone on FXO?
At 07:09 AM 01/29/2006, you wrote: I just tried that, but it doesn't help. There is still one ring tone produced before asterisk executes Answer(). And there is no caller ID being forwarded to the destination channel, which actually I need. That is why I have usecallerid set to yes. I added a wait(2) as the first item in my incoming dial plan and that seemed to increase my chances of picking up the CID on a TDM400 ZAP channel by a significant amount. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HandyTone 488 ata?
On Sun, 2006-01-29 at 12:36 -0600, Rich Adamson wrote: Anyone tried to muck around with using the 488 for both fxs and fxo with asterisk? I've been playing with one for the last couple of days, and it looks like its a little lower quality then the spa3k. No gain settings, echo canceller is less then ideal on long analog pstn loops, etc. Anyone with good experiences? I played with FXO on the HT488 a bit, but didn't have a whole lot of luck. We had a bit of a problem with echo, but more seriously the thing kept getting itself into a variety of wedged states: sometimes it would lock up altogether (usually with its button lit up), and sometimes it would refuse to auto-answer calls coming in on the FXO interface. These latter problems have been severe enough that I didn't bother trying to diagnose the echo thing. Plus, even when set to auto answer after 1 ring, it often seemed to wait for three or four rings before picking up. Our HT386s are also a little bit prone to locking up and needing to be rebooted, but that seems to be a different problem: it occurs less often than on the HT488, and seems to be triggered by something to do with call transfers (which we never did with the 488). I've just bought an SPA-3000 to replace the HT488, though I haven't installed it yet. I'm hoping that I'll have a better experience with this one. If that works out, I might toss the 386s in favour of SPA-2000s as well. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Sounds like maybe a firewall is involved somewhere. Are you sure there are none in the path (including on your asterisk box)? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HandyTone 488 ata?
Anyone tried to muck around with using the 488 for both fxs and fxo with asterisk? I've been playing with one for the last couple of days, and it looks like its a little lower quality then the spa3k. No gain settings, echo canceller is less then ideal on long analog pstn loops, etc. Anyone with good experiences? I played with FXO on the HT488 a bit, but didn't have a whole lot of luck. We had a bit of a problem with echo, but more seriously the thing kept getting itself into a variety of wedged states: sometimes it would lock up altogether (usually with its button lit up), and sometimes it would refuse to auto-answer calls coming in on the FXO interface. These latter problems have been severe enough that I didn't bother trying to diagnose the echo thing. Plus, even when set to auto answer after 1 ring, it often seemed to wait for three or four rings before picking up. Our HT386s are also a little bit prone to locking up and needing to be rebooted, but that seems to be a different problem: it occurs less often than on the HT488, and seems to be triggered by something to do with call transfers (which we never did with the 488). I've just bought an SPA-3000 to replace the HT488, though I haven't installed it yet. I'm hoping that I'll have a better experience with this one. If that works out, I might toss the 386s in favour of SPA-2000s as well. Thanks for the response. I haven't hit the lockup issue as yet, but then the 488 has only been around here for about 24 hours. Echo is considerably worse then the spa3k. Seems the spa3k functions pretty well (had a few since they first came out), but the echo can on long analog loops leaves some to be desired as well. Short loops seem to work just fine. Rich ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New C7960 won't tftp?
Just received a new Cisco 7960 (not refurb, but brand new) and it won't tftp the initial config file (OS79XX.TXT) from an FC3 box. The 7960 does get an appropriate dhcp response including the tftp address. Using a sniffer, I see the tftp request being sent from the 7960 to the FC3 box, but the FC3 box responds with error code 4 (Illegal TFTP opertion) and an error message of Request not null-terminated. The tftp request (as observed with the sniffer) does indicate the initial request is null terminated without a shadow of a doubt. I even rebooted the FC3 (hated to do that since its uptime was very good) with exactly the same response. Checked the permissions, etc, and they all look fine. Other Cisco 7960's have been booting just fine using this same FC3 tftp server. Anyone have any thoughts on possible things to look at? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Moprobe Zaptel error
So, if anybody is interested, i think i got it! i executed in the linux kernel source directory 'menu xconfig' and the graphical config windows appears (nicer to me) then in processor type i changed from PentiumPro (why was this value here?) to K6, but after recompiling zaptel and modprobing: that didn't work (but it is a K6II 3Dprocessor) So i set again this option, to the value 586then in zaptel, make clean, make linux26 and make install Then when modprobe zaptel no errors appeared bye, Maxi 2006/1/29, Maxi Belino [EMAIL PROTECTED]: yes, i do have /boot/config so then? what should i do? thanks again. Maxi2006/1/29, Tzafrir Cohen [EMAIL PROTECTED]: On Sun, Jan 29, 2006 at 01:46:32PM +, Maxi Belino wrote: Hi, i'm trying to install a compatible modem to act as a X100P, and i would appreciate some help here, this is what is hapening when i modprobe zaptel: # modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.11-6mdk-i586-up-1GB/misc/zaptel.ko): Invalid module format firstly, i did: 'make clean' then 'make linux26' and then 'make install' into zaptel directory Also, in the Linux kernel source Makefile, i replaced: EXTRAVERSION = -6mdkcustom with this: EXTRAVERSION = -6mdk-i586-up-1GB See the error in /var/log/messages: Jan 29 10:41:46 maxilinux kernel: zaptel: version magic ' 2.6.11-6mdk-i586-up-1GB 686 gcc-3.4' should be '2.6.11-6mdk-i586-up-1GB 586 gcc-3.4' Apparently the difference comes from '686' when it should be '586' How do i fix this? (i've been reading but i did not find the answer to this) To give you some info: # uname -r 2.6.11-6mdk-i586-up-1GB # uname -a Linux maxilinux 2.6.11-6mdk-i586-up-1GB #1 Tue Mar 22 15:46:07 CET 2005 i586 AMD-K6(tm) 3D processor unknown GNU/LinuxI don't know how current Mandriva kernel packages are layed, but youshould be able to find the .config file of your existing kernel in that package. Maybe under /usr/share/doc , or maybe in /boot/config* .--Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il| | a Mutt's[EMAIL PROTECTED] | |bestICQ# 16849755 | | friend___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HandyTone 488 ata?
On Sun, 2006-01-29 at 13:24 -0600, Rich Adamson wrote: Seems the spa3k functions pretty well (had a few since they first came out), but the echo can on long analog loops leaves some to be desired as well. Short loops seem to work just fine. Thanks for the information. Sounds encouraging. The only significant feature that the SPAs seems to be missing compared to the HTs is the Early Dial thing (where it sends each digit to Asterisk until it gets something other than a 484 response back). Without that, my users need to either wait for a timeout or dial #, neither of which is terribly appealing. However, I haven't ever quite succeeded in making Early Dial work reliably on the HTs in any case (although it seems to be fine on our GXP-2000s), so I guess it's not as much of a loss as all that. And, well, I guess I can live in hope that Sipura will implement this one day. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
No Firewalls involved, the test has been simplified down to two sip phones on a LAN and still no audio. For waht it's worth IAX2 still works fine. Steve - Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Sounds like maybe a firewall is involved somewhere. Are you sure there are none in the path (including on your asterisk box)? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HandyTone 488 ata?
On Jan 29, 2006, at 11:24 AM, Rich Adamson wrote: Anyone tried to muck around with using the 488 for both fxs and fxo with asterisk? I've been playing with one for the last couple of days, and it looks like its a little lower quality then the spa3k. No gain settings, echo canceller is less then ideal on long analog pstn loops, etc. Thanks a lot for posting this topic! I was kind of wondering along the same lines, but didn't have the whole thing together in my mind. I am a newb, so my confidence is lacking I have been using the HT-488 for several months now. My observations are as follows. The locking up that is described by the other poster, seems to be to be related to the routing functions of this device. I have gotten up to 20 days of uptime (current), by disabling this and using the WAN port only. I don't need it's routing anyhow. This issues with the echo and the volume are the same here. I sometimes here echo, and then in the midst of the call I hear a loud static burst and the echo stops. I wasn't entirely sure why this was happening, but it happens both when using the local FXS, and when using a remote AG-168V, so it's clearly and FXO issue. I have a gut feeling that a gain adjustment for this device might also help it's echo cancel function? Is there anyway that anyone knows of to adjust the gain? I am going to NEED to replace this device with some other FXO (external), so if there are other superior options I would love to hear them. I know about the Sipura 3000, but have heard about some issues with that too... Thanks, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
Have you tried increasing the debug level and watching the cli? No Firewalls involved, the test has been simplified down to two sip phones on a LAN and still no audio. For waht it's worth IAX2 still works fine. Steve - Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Sounds like maybe a firewall is involved somewhere. Are you sure there are none in the path (including on your asterisk box)? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
On Jan 29, 2006, at 10:30 AM, Warren Burstein wrote: I took a look at the asterisk-1.2.3 Makefile, seems to me that the WARNING is just a list of all the .so files found in the modules directory that aren't also found in a subdirectory, it isn't checking that they were built with the current version. So it's going to complain about the modules that come from asterisk-addons every time make install is run in asterisk, no matter what. Not a big problem once you learn to ignore the message, but people are probably going to keep asking what it means. Actually in this case, it means it! I updated from 1.21 to current as of yesterday, and my * wouldn't start as it complained about the modules it couldn't load. I removed the following to get it starting up again: app_enumlookup.so app_groupcount.so app_md5.so app_txtcidname.so func_cut.so This seems to be ok, but I wonder if I have broken anything with this approach? I didn't understand that removing all the modules and doing a make install would replace the needed ones? What about the g729 codecs aren't those stored here too? Thanks, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wifi phone set-up
I got some troubles with my wifi phone. What phone is this? Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wifi phone set-up
Outbound proxy Proxy IP stun01.sipphone.com Port:: 3478 STUN servers are not outbound SIP proxies. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
On 13:09, Sun 29 Jan 06, Martin Joseph wrote: I removed the following to get it starting up again: app_enumlookup.so app_groupcount.so app_md5.so app_txtcidname.so func_cut.so Both the README and the UPGRADE listed that those functions became obsolete and were replaced by dialplan functions. It got me too the first time, but after reading some more docs and the release messages all got clear to me. Now everytime I upgrade I first move away the module/app/func dir just to be sure. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New C7960 won't tftp?
Disregard... should have been smarter and looked at the wiki. Damn! http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx Just received a new Cisco 7960 (not refurb, but brand new) and it won't tftp the initial config file (OS79XX.TXT) from an FC3 box. The 7960 does get an appropriate dhcp response including the tftp address. Using a sniffer, I see the tftp request being sent from the 7960 to the FC3 box, but the FC3 box responds with error code 4 (Illegal TFTP opertion) and an error message of Request not null-terminated. The tftp request (as observed with the sniffer) does indicate the initial request is null terminated without a shadow of a doubt. I even rebooted the FC3 (hated to do that since its uptime was very good) with exactly the same response. Checked the permissions, etc, and they all look fine. Other Cisco 7960's have been booting just fine using this same FC3 tftp server. Anyone have any thoughts on possible things to look at? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco VG200 as FXO for * ?
Anyone used a Cisco VG200 as FXO gateway for * ? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wifi phone set-up
Nabeel Jafferali wrote: I got some troubles with my wifi phone. What phone is this? pulver phone ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Access Codes
I would like to setup Asterisk as follows: When users make inter-office calls they can dial the extensions, however if they want to make an external call, that they enter a code on their phone, before they external call can go through. We would like to give each user an access code, this way we can limit certain employees from making certain calls to certain places. What's the best recommendation for this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wifi phone set-up
Throw it in the trash now. There's next to no support for these. No firmware upgrades. The are VERY SLOOW in responding to network calls too. All in all not a very astute purchase. I should know; I've had 5 of them. I use the UTStarcom F1000 currently. Much better but still not good. Mark Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Ronald Wiplinger wrote: Nabeel Jafferali wrote: I got some troubles with my wifi phone. What phone is this? pulver phone ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wifi phone set-up
On 17:26, Sun 29 Jan 06, Mark Phillips wrote: Throw it in the trash now. There's next to no support for these. No firmware upgrades. The are VERY SLOOW in responding to network calls too. All in all not a very astute purchase. I should know; I've had 5 of them. I use the UTStarcom F1000 currently. Much better but still not good. What about the Cisco 7920 or the kirk ip600? I know the ip600 and it works like a charm, even with your old dect/GAP phones. The cisco 7920 looks great too, and what I've heard it performs good too. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Access Codes
On Sun, 2006-01-29 at 18:00 -0400, Dakota wrote: I would like to setup Asterisk as follows: When users make inter-office calls they can dial the extensions, however if they want to make an external call, that they enter a code on their phone, before they external call can go through. We would like to give each user an access code, this way we can limit certain employees from making certain calls to certain places. What's the best recommendation for this? There are 2 basic ways to do this. You can in your dialplan do a read and presumably a db check for the extension (I assume you dont want hardcoded accounts, but ...) The other way is via an agi that does the same thing. AGIs generally take more cpu than equivalent dialplan entries for short lived applications, however for a long lived agi it may not (although it can take more ram). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk + XEN does it make sense?
Hi List, I was wondering if anybody had tried running Asterisk inside virtualization software such as Xen. Are there known problems doing it? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wifi phone set-up
Mark Phillips wrote: Throw it in the trash now. There's next to no support for these. No firmware upgrades. The are VERY SLOOW in responding to network calls too. Thanks for your suggestion, but it still did not explain how to set-up! I figured out that if I set outbound proxy same as proxy it works again. However Asterisk complains: [Jan 30 06:36:18] -- Got SIP response 400 Bad Request back from 203.70.36.126 [Jan 30 06:37:02] -- Got SIP response 400 Bad Request back from 203.70.36.126 [Jan 30 06:37:57] -- Got SIP response 400 Bad Request back from 203.70.36.126 [Jan 30 06:38:52] -- Got SIP response 400 Bad Request back from 203.70.36.126 What is that? Can it be stopped? It was one of my most expensive phones (including shipment to Taiwan). I will get new one - made in Taiwan. If you like I can offer you. Even this pulver wisip is bad, it can still be used as a good reference. bye Ronald Wiplinger All in all not a very astute purchase. I should know; I've had 5 of them. I use the UTStarcom F1000 currently. Much better but still not good. Mark Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Ronald Wiplinger wrote: Nabeel Jafferali wrote: I got some troubles with my wifi phone. What phone is this? pulver phone ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Access Codes
Dakota a écrit : I would like to setup Asterisk as follows: When users make inter-office calls they can dial the extensions, however if they want to make an external call, that they enter a code on their phone, before they external call can go through. We would like to give each user an access code, this way we can limit certain employees from making certain calls to certain places. What's the best recommendation for this? It depends You could create an context for each employee with proper permissions. Then they wouldn't be able to place calls they are not supposed to unless they were using somebody elses'. If you wanted to, you could have an extension (effectively making this special extension the password) in each context which jumps to the right context (in which you could play a dialtone, grab some digits and dial the outside number). But maybe this is too restrictive. Have you considered telling your employees you are not normally allowed to ring such and such number and tell them that their calls are monitored (through CDR checking for example)? This way, you could write a script which would raise an alert for any unauthorized calls. In my view, such a system would be better because in rare instances, employees might have a good reason to place a call they are not normally allowed to place. Nothing sucks more than a system which goes b. Can't do that when you actually have a good reason for doing 'that'... Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Access Codes
Or you can use authenticate() and have it take its 'passwords' form a text file on your machine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Sunday, January 29, 2006 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Access Codes On Sun, 2006-01-29 at 18:00 -0400, Dakota wrote: I would like to setup Asterisk as follows: When users make inter-office calls they can dial the extensions, however if they want to make an external call, that they enter a code on their phone, before they external call can go through. We would like to give each user an access code, this way we can limit certain employees from making certain calls to certain places. What's the best recommendation for this? There are 2 basic ways to do this. You can in your dialplan do a read and presumably a db check for the extension (I assume you dont want hardcoded accounts, but ...) The other way is via an agi that does the same thing. AGIs generally take more cpu than equivalent dialplan entries for short lived applications, however for a long lived agi it may not (although it can take more ram). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wildcard matching in dialplan
On Sun, 2006-01-22 at 18:18 +0100, Wilson Pickett wrote: You could also use a trick like *21* going to a new context and waiting for digits (with a slighly longer timeout) and have it trigger on the longest possible number. perhaps if local extension were of the form 2nnn or 2nn and you want to use both local and normal local POTS numbers you can use two or more extensions: _*21*2xx* _*21*2xxx* _*21*nxxn* etc. Use the include= trick to prioritize the last three properly. Thanks for the suggestions. Unfortunately your second option wouldn't have worked for me because my users want to forward calls to all manner of external numbers, including international ones with unpredictable formats. I thought about doing the first thing you mentioned, but I wasn't sure whether this would work if someone programmed a redirect into a speed-dial key and Asterisk got hit with the whole number at once. Anyway, I ended up hacking the pattern matching code in pbx.c to support the kind of patterns that I needed. It's a bit gruesome, but seems to be working well enough. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agi debug - unable to set normal priority
In my agi-debug i get the following error-message: AGI Rx Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority: I have the same problem with all (shell) AGIs. Not sure when it started (about two days ago) and why, i tried to restart asterisk and my server and also reinstalling asterisk. I'm currently using bristuff-Asterisk 1.2.2 (with no audio patch) on Debian Sarge with a 2.6.14-2-686 Kernel (backports.org), run as non-root. The AGIs have worked with this setup. I've tested a Debian Sarge with Asterisk 1.2.3, same Kernel, also as non-root, but on a different computer with wmware and there it worked. BTW: The priority set is called in res/res_agi.c (i guess): static int launch_script(...) /* Don't run AGI scripts with realtime priority -- it causes audio stutter */ ast_set_priority(0); /* Execute script */ execv(script, argv); This was introduced with http://bugs.digium.com/view.php?id=4930 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HandyTone 488 ata?
We started out useing SPA2k but they were prone to stop talking to the ethernet. OK after reboot for awhile but cannot keep going to customer sites and rebooting things. switched to spa2001 and somewhat better but they keept losing registrations and then could not talk to them remotely. Again the reboot issue. Then switched to 386 and the couple I tried worked ok so then installed several 286. Bad move. The 286 regularly lose registrations and lock up. Have beugun replacing them with some 386. so far other than a doa my best luck is with the 386, Of course I do have one that is working great, but I can no longer talk to it:( Will have to replace next time I get onsite with that customer. On Jan 29, 2006, at 1:15 PM, Phil Blundell wrote: On Sun, 2006-01-29 at 12:36 -0600, Rich Adamson wrote: Anyone tried to muck around with using the 488 for both fxs and fxo with asterisk? I've been playing with one for the last couple of days, and it looks like its a little lower quality then the spa3k. No gain settings, echo canceller is less then ideal on long analog pstn loops, etc. Anyone with good experiences? I played with FXO on the HT488 a bit, but didn't have a whole lot of luck. We had a bit of a problem with echo, but more seriously the thing kept getting itself into a variety of wedged states: sometimes it would lock up altogether (usually with its button lit up), and sometimes it would refuse to auto-answer calls coming in on the FXO interface. These latter problems have been severe enough that I didn't bother trying to diagnose the echo thing. Plus, even when set to auto answer after 1 ring, it often seemed to wait for three or four rings before picking up. Our HT386s are also a little bit prone to locking up and needing to be rebooted, but that seems to be a different problem: it occurs less often than on the HT488, and seems to be triggered by something to do with call transfers (which we never did with the 488). I've just bought an SPA-3000 to replace the HT488, though I haven't installed it yet. I'm hoping that I'll have a better experience with this one. If that works out, I might toss the 386s in favour of SPA-2000s as well. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] simulating a few thousand SIP clients?
Hello Roy, Have you heard of Sipp? http://sipp.sourceforge.net/. I am pretty sure it can do what you desire. Also a commercial tool from Empirix, Hammer NXT. (http://www.empirix.com/default.asp?action=articleID=64) Cheers, Omar On 1/29/06, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: sure, but I need to simulate the SIP REGISTER and OPTION traffic sent by ATAs as well. Best regards Roy Sigurd Karlsbakk [EMAIL PROTECTED] --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. On Jan 29, 2006, at 7:26 PM, Wai Wu wrote: Set up another * and use the manager api to make lots of calls to the other one. You can even make hundresd calls at a time. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Roy Sigurd Karlsbakk Sent: Sunday, January 29, 2006 1:19 PM To: Asterisk Non-Commercial Discu Subject: [Asterisk-Users] simulating a few thousand SIP clients? hi i'm setting up a rig to handle quite a few SIP clients, so i need a way to simulate, say, 20k SIP ATAs. Does anyone know how? This should of course be as close as possible to 'reality', meaning n% calls per client and the usual REGISTER/OPTION traffic. thanks Best regards Roy Sigurd Karlsbakk [EMAIL PROTECTED] --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agi debug - unable to set normal priority
Carsten Bock wrote: In my agi-debug i get the following error-message: AGI Rx Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority: Oups there something missing, the complete error message is AGI Rx Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority: Unable to set normal priority Posted in the original mail from Christian http://lists.digium.com/pipermail/asterisk-users/2006-January/142564.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Access Codes
Dakota wrote: I would like to setup Asterisk as follows: When users make inter-office calls they can dial the extensions, however if they want to make an external call, that they enter a code on their phone, before they external call can go through. We would like to give each user an access code, this way we can limit certain employees from making certain calls to certain places. What's the best recommendation for this? Use astcc (prepaid card system) with pin! You can give the rates all to free if you want. Give each one who is allowed a balance, if they do not have, they cannot call. You also would have a nice statistic for each call, bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + XEN does it make sense?
Jean-Michel Hiver wrote: Hi List, I was wondering if anybody had tried running Asterisk inside virtualization software such as Xen. Are there known problems doing it? Cheers, Jean-Michel. I have been running several asterisk xen servers for a few months now. Problems would depend on what you plan to do. I do know that I have tested it with 6 channels g729 transcoded on a 96mb xen virtual and it still worked. I do all my agi development and testing on virtuals. It's so easy to build and rebuild test servers this way. It also provides an easy way to revert to a snapshot. If you suspect that 1.2.x has a new bug you can quickly switch back to the previous setup to verify that. I should also mention that I am using xen for some client ecommerce stuff. It provides a means where we can have redundancy at affordable costs. I have virtual servers spread over 3 data centers this way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Access Codes
Can I get some more information on this? Are there any drawbacks? - Original Message - From: Alexander Lopez [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 29, 2006 6:58 PM Subject: RE: [Asterisk-Users] Access Codes Or you can use authenticate() and have it take its 'passwords' form a text file on your machine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Sunday, January 29, 2006 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Access Codes On Sun, 2006-01-29 at 18:00 -0400, Dakota wrote: I would like to setup Asterisk as follows: When users make inter-office calls they can dial the extensions, however if they want to make an external call, that they enter a code on their phone, before they external call can go through. We would like to give each user an access code, this way we can limit certain employees from making certain calls to certain places. What's the best recommendation for this? There are 2 basic ways to do this. You can in your dialplan do a read and presumably a db check for the extension (I assume you dont want hardcoded accounts, but ...) The other way is via an agi that does the same thing. AGIs generally take more cpu than equivalent dialplan entries for short lived applications, however for a long lived agi it may not (although it can take more ram). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Access Codes
Drawbacks are few in my opinion. The onlys issue is that users will hear
'Please enter your password They get three attempts and if they do not
enter it right the system goes to priority + 101.
Example:
Exten = _91X.,1,Authenticate(/etc/asterisk/ldusers.txt)
Exten = _91X.,2,Dial(Zap/g2/${EXTEN:1})
Exten = _91X.,3,Congestion
Exten = _91X.,102,Congestion
The ldusers.txt file would have on entry per line:
02345
43535
89033
23903
Etc.
We use 9 as a prefix for outside lines your setup may vary...
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dakota
Sent: Sunday, January 29, 2006 7:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Access Codes
Can I get some more information on this?
Are there any drawbacks?
- Original Message -
From: Alexander Lopez [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, January 29, 2006 6:58 PM
Subject: RE: [Asterisk-Users] Access Codes
Or you can use authenticate() and have it take its 'passwords' form a
text file on your machine.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
trixter aka Bret McDanel
Sent: Sunday, January 29, 2006 5:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Access Codes
On Sun, 2006-01-29 at 18:00 -0400, Dakota wrote:
I would like to setup Asterisk as follows:
When users make inter-office calls they can dial the extensions,
however if they want to make an external call, that they
enter a code
on their phone, before they external call can go through.
We would like to give each user an access code, this way we
can limit
certain employees from making certain calls to certain places.
What's the best recommendation for this?
There are 2 basic ways to do this. You can in your dialplan
do a read and presumably a db check for the extension (I
assume you dont want hardcoded accounts, but ...)
The other way is via an agi that does the same thing. AGIs
generally take more cpu than equivalent dialplan entries for
short lived applications, however for a long lived agi it may
not (although it can take more ram).
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wifi phone set-up
Mark - The new UTStarCom F3000 should be shipping soon. I have done a bit of preliminary testing and it seems to work very well. Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Mark Phillips [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 29, 2006 5:26 PM Subject: Re: [Asterisk-Users] Wifi phone set-up Throw it in the trash now. There's next to no support for these. No firmware upgrades. The are VERY SLOOW in responding to network calls too. All in all not a very astute purchase. I should know; I've had 5 of them. I use the UTStarcom F1000 currently. Much better but still not good. Mark Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Ronald Wiplinger wrote: Nabeel Jafferali wrote: I got some troubles with my wifi phone. What phone is this? pulver phone ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Access Codes
On Sun, 2006-01-29 at 19:58 -0500, Alexander Lopez wrote: Drawbacks are few in my opinion. The onlys issue is that users will hear 'Please enter your password They get three attempts and if they do not enter it right the system goes to priority + 101. An other drawback in my opinion is one I originally said in my email that I was corrected on ... That its a flat text file which limits dynamic passwords, in that you have to either write something that will allow users to change their passwords, which is more difficult (although not impossible, just watch file locking and permission issues) than using a DB. The other alternative is that an admin type has to change the passwords for the individuals. While many installations dont care as much as I aparently do over passwords, this can still be a cumbersome task in a large installation to do them manually like that. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web interface
I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to get IP of eth0
I would set up the softphone on a public address and see if it works first. How do you set up the sip.conf for the softphone? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of SoFieSent: Sunday, January 29, 2006 1:41 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Unable to get IP of eth0 Hi all, Im trying to set up my asterisk server, but Im having a few problems. My server is running with a public IP address. When I want to set up a call with a softphone in my private network behind a router Im always having an error message. In the CLI session we get a message when the softphone starts up. But after that we get immediately the message Unable to get IP of eth0 : cannot assign requested addresses. Im using NAT so I changed in my router the port forwarding. So the SIP/RTP ports are forwarding to my PC and theres just one client in my network. Can somebody help me? Thanks, Sofie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] simulating a few thousand SIP clients?
I see. But are you going to setup a few thousand entries in the sip.conf, one for each of ATA? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Roy Sigurd Karlsbakk Sent: Sunday, January 29, 2006 1:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] simulating a few thousand SIP clients? sure, but I need to simulate the SIP REGISTER and OPTION traffic sent by ATAs as well. Best regards Roy Sigurd Karlsbakk [EMAIL PROTECTED] --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. On Jan 29, 2006, at 7:26 PM, Wai Wu wrote: Set up another * and use the manager api to make lots of calls to the other one. You can even make hundresd calls at a time. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Roy Sigurd Karlsbakk Sent: Sunday, January 29, 2006 1:19 PM To: Asterisk Non-Commercial Discu Subject: [Asterisk-Users] simulating a few thousand SIP clients? hi i'm setting up a rig to handle quite a few SIP clients, so i need a way to simulate, say, 20k SIP ATAs. Does anyone know how? This should of course be as close as possible to 'reality', meaning n% calls per client and the usual REGISTER/OPTION traffic. thanks Best regards Roy Sigurd Karlsbakk [EMAIL PROTECTED] --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: How to remove first ring tone on FXO?
Aryanto Rachmad wrote: Thanks a lot Dean, I think there is a way to remove that ring tone and also still have the caller ID from the incoming call. I have been trying to find that on zaptel.c, chan_zap.c and pbx.c, but I could not find that. Could you please let me know which part of the codes handling that? I don't see how this can be done. The first ringback tone is sent from the serving telco CO, not from asterisk. Sequence is: 1. CO sends one ringing pulse to the called party. This both generates an audible ring and wakes up the caller-ID modem to listen for CLID. Simultaneously, ringback tone is played to the calling party. 2. CO sends modem burst with caller-ID information. 3a. Call is answered by called party going off-hook and drawing loop current. Supervision occurs and audio is cut through. 3b. Call is not answered. Ringing voltage is sent every six seconds as well as ringback tone to calling party until the calling party abandons the call or CO times out and releases. In a loop-start FXO scenario the caller will hear the first ringback which is generated by the telco switch in advance of your receiving CLID. ISDN (BRI or PRI) will allow you to receive CLID in advance of audible ringing or ringback to the caller, but you aren't going to be able to do this with FXO. -- Jay Hennigan - CCIE #7880 - Network Administration - [EMAIL PROTECTED] NetLojix Communications, Inc. - http://www.netlojix.com/ WestNet: Connecting you to the planet. 805 884-6323 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
On Jan 29, 2006, at 1:24 PM, Michiel van Baak wrote: On 13:09, Sun 29 Jan 06, Martin Joseph wrote: I removed the following to get it starting up again: app_enumlookup.so app_groupcount.so app_md5.so app_txtcidname.so func_cut.so Both the README and the UPGRADE listed that those functions became obsolete and were replaced by dialplan functions. It got me too the first time, but after reading some more docs and the release messages all got clear to me. Now everytime I upgrade I first move away the module/app/func dir just to be sure. Thanks, duh! I guess RTFM is the answer... Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web interface
AMP hands down is STILL the best... though a few are catching up quickly On Mon, 2006-01-30 at 01:29 +, Strain Jer wrote: I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web interface
On Mon, 2006-01-30 at 10:42 +0800, Lists wrote: AMP hands down is STILL the best... though a few are catching up quickly The best non-web interface (I had considered copying it into a webinterface) is the cocoa app for the mac asterisk stuff. There is a reason you dont really see questions about it. http://www.astmasters.net somewhere there are screenshots, the interface is slick and takes virtually no telephony knowledge to set something up. Converted to a web UI would potentially make it a lot better :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
Yes I have. I have been battling this issue since wednesday 1-25 And so far have tried many things. Have also tried RTP debug and do not see ANY RTP when the call is made. I will keep working at this until I figure it out but right now am very stumped and frusterated. The software update SHOULD have fixed it as it has for many others. Steve Have you tried increasing the debug level and watching the cli? No Firewalls involved, the test has been simplified down to two sip phones on a LAN and still no audio. For waht it's worth IAX2 still works fine. Steve - Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Sounds like maybe a firewall is involved somewhere. Are you sure there are none in the path (including on your asterisk box)? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
Steve: I'm picking up the tail end of a thread, so apologies if this is offtrack... Have you perhaps got an old set of EXECUTABLES in your path, that are being picked up before your newly compiled ones? Roger Steve Gladden wrote: Yes I have. I have been battling this issue since wednesday 1-25 And so far have tried many things. Have also tried RTP debug and do not see ANY RTP when the call is made. I will keep working at this until I figure it out but right now am very stumped and frusterated. The software update SHOULD have fixed it as it has for many others. Steve Have you tried increasing the debug level and watching the cli? No Firewalls involved, the test has been simplified down to two sip phones on a LAN and still no audio. For waht it's worth IAX2 still works fine. Steve - Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Sounds like maybe a firewall is involved somewhere. Are you sure there are none in the path (including on your asterisk box)? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic / Voip Forum
Hi everyone, www.dialogictrader.com dialogic and general voip hardware forum This is no way a plug but I wanted some user feedback on a site I had put together which allows people that use voip and dialogic hardware to come together. They do not necessarily have to be used together but I figured why not. The site is on a dev site so feel free to check it out. I do not plan to make money from this but I got tired of admins hijacking good deals and I want to make this site completely user driven. Many people come across old phone and voicemail boxes and I just figured why not create a place outside of ebay for them to be discussed and or sold. I know its odd because dialogic is specific and voip is very broad but I use dialogic stuff every day and I want to kill 2 birds with one stone. J Any advice would be appreciated ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer (SIP REFER) - AccountCode not available?
I have a snom 320 connected to an Asterisk server. I do some weird things
using the AccountCode as an identifier. When the snom makes a call, the
AccountCode is used successfully in the dialplan as a variable
${ACCOUNTCODE}.
When that same call is transferred using the button on the snom, I see a SIP
REFER message being received on the * server and the call is transferred -
however, this new call, when going through the steps of the dialplan, has
a blank AccountCode.
The transferred call is initiated in the correct context, meaning Asterisk
is treating it as a call from the correct user, however it seems to forget
the user's AccountCode (which is set in sip.conf).
Any ideas?
Nabeel
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
Have you verified that you are actually sending sound over the RTP streams? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith Sent: Friday, January 27, 2006 11:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question What's you mix of calls going SIP/IAXand to PSTN? We've done some benchmark experiments on a 3GHz HT box with 1GB of ram, mirrored traditional IDE disks. The box has a Digium quad-PRI, a TDM40B, a TDM22B and a Sirrix quad-BRI board in it. This box can run 120 active calls over 4 PRI spans. Its running MusicOnHold into 60 of the channels, playing various GSM prompts into the other 60. The user cpu usage is about 25%, the system cpu about 25% also. We can add to that 5000 registered SIP peers and 5000 registered IAX2 peers - total of about 100 registration refreshes per second. That adds about 40% more user CPU and pretty much fills up CPU. Audio quality is still perfectly fine, and PRI slips few and far between. Load average for the whole mix sits between 5 and 10. About 550Kb/s out and 400Kb/s out on the ethernet for the registration traffic. Also on www.voip-info.org - search for dimensioning Rob On 1/28/06, Vic [EMAIL PROTECTED] wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTPP
Hi Sir, My problem is when I click on pricelist, i have an error there's something wrong on the pricelist database. When I looked at the database and search for a table called pricelist there's nothing there. I foolowed the querires on the the structure but also found any query that creates the pricelist table. Is the pricelist going to be created at the start or after I've setup everything? Thank You Regards, Ronald JP Carballo wrote: Under Rates click on - Pricelists then Add... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
Signate sells a single server that can get you to the call volumes you need. Paul Mahler [EMAIL PROTECTED] www.signate.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vic Sent: Saturday, January 28, 2006 7:16 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. Or do 30 MHz are only necessary for transcoding? In other words, if it comes in as SIP and we keep it that way, canwe make ita bt more feasible number? Zoa [EMAIL PROTECTED] wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialing 2 channels at the same time with different callerID number?
Can anyone think of a way to dial 2 different numbers at the same time, but set the callerID number differently for each channel? The application is a simultaneous ring of an office extension and a cell phone where the user wants to know that the call to the cell phone is a redirected call from the office. Maybe the use of a local channel with changed caller ID behavior as the second channel dialed? To add additional complexity, I need to be able to pass an argument to a macro to enable/disable this feature TIA Damon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialing 2 channels at the same time with differentcallerID number?
You can then call up the macro like this: [extensions] Exten = 1120,1,Macro(call-cell,SIP/120,Local/[EMAIL PROTECTED]) Exten - 2120,1,Macro(call-cell,SIP/120) [macro-call-cell] Exten = s,1,Dial($ARG1ARG2) [cellulars] Exten = 120,1,Set(CALLERID(num)=551212) Exten = 120,2,Dial(Zap/g2/8002MYCELL) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Sunday, January 29, 2006 11:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] dialing 2 channels at the same time with differentcallerID number? Can anyone think of a way to dial 2 different numbers at the same time, but set the callerID number differently for each channel? The application is a simultaneous ring of an office extension and a cell phone where the user wants to know that the call to the cell phone is a redirected call from the office. Maybe the use of a local channel with changed caller ID behavior as the second channel dialed? To add additional complexity, I need to be able to pass an argument to a macro to enable/disable this feature TIA Damon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTPP
This doesn't really belong on the asterisk-users list. ASTPP has it's own mailing list. This can be found @ www.astpp.org. I, or someone else will be happy to help you either there or on the forums. On your 1st post please mention what version of ASTPP you are using. Thanks, Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.astpp.org Ronald Ramos wrote: Hi Sir, My problem is when I click on pricelist, i have an error there's something wrong on the pricelist database. When I looked at the database and search for a table called pricelist there's nothing there. I foolowed the querires on the the structure but also found any query that creates the pricelist table. Is the pricelist going to be created at the start or after I've setup everything? Thank You Regards, Ronald JP Carballo wrote: Under Rates click on - Pricelists then Add... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk + XEN does it make sense?
I use VMWare, but will start testing XEN...I use VMWare to slice up some nice big servers to provide dedicated hosted PBXes. We also use the VMs for easy deployment and is a vital part of our DR Plan... Now, we are full VoIP...not T1 or PRI cards... -Original Message- From: Paul [mailto:[EMAIL PROTECTED] Sent: Sunday, January 29, 2006 7:17 PM To: Asterisk User List Subject: Re: [Asterisk-Users] Asterisk + XEN does it make sense? Jean-Michel Hiver wrote: Hi List, I was wondering if anybody had tried running Asterisk inside virtualization software such as Xen. Are there known problems doing it? Cheers, Jean-Michel. I have been running several asterisk xen servers for a few months now. Problems would depend on what you plan to do. I do know that I have tested it with 6 channels g729 transcoded on a 96mb xen virtual and it still worked. I do all my agi development and testing on virtuals. It's so easy to build and rebuild test servers this way. It also provides an easy way to revert to a snapshot. If you suspect that 1.2.x has a new bug you can quickly switch back to the previous setup to verify that. I should also mention that I am using xen for some client ecommerce stuff. It provides a means where we can have redundancy at affordable costs. I have virtual servers spread over 3 data centers this way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
On Fri, Jan 27, 2006 at 03:20:16PM -0600, Dan Littlejohn wrote: I was confused about the modules. Got this warning when upgrading to 1.2.3 even when using the most current asterisk-addons and even svn asterisk-addons. WARNING WARNING WARNING Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. [ order of modules changed a bit, if you don't mind ] app_addon_sql_mysql.so cdr_addon_mysql.so res_config_mysql.so format_mp3.so Those are from asterisk-addons . app_rxfax.so app_txfax.so Those are from the apps of app_saycountpl.so app_striplsd.so app_substring.so Standard asterisk modules, IIRC. See note below. chan_modem_aopen.so chan_modem_bestdata.so chan_modem_i4l.so chan_modem.so chan_modem* may be leftovers from older asterisk 1.0.x ? They are now not built by defaults. Look at the dates. WARNING WARNING WARNING Do not understand how to fix this? Do not know if that would also be related to the ops crashing. Remove modules you don't need anymore and try again :-) Theoretically you should not be required to rebuild modules on a minor upgrade (that is: as long as it remains 1.2). But if there are older leftovers, they need removing. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
On Fri, Jan 27, 2006 at 04:03:23PM -0600, Joseph Tanner wrote: Quick and dirty solution: mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules.bak And just as a reminder: those are basically exactly the problems package management systems are here to solve. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Moprobe Zaptel error
On Sun, Jan 29, 2006 at 06:54:05PM +, Maxi Belino wrote: yes, i do have /boot/config so then? what should i do? if you build a custom kernel, no problems. But if you use a distro kernel, you should point the makefile to the directory where your kernel source is (or at least: suffucuent headers for a build). This tree should have the configuration of the kernel you currently run [actually: of the kernel you build the modules for. But I assume that in your case those two are the same]. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Access Codes
On Sun, Jan 29, 2006 at 05:16:28PM -0800, trixter aka Bret McDanel wrote: An other drawback in my opinion is one I originally said in my email that I was corrected on ... That its a flat text file which limits dynamic passwords, in that you have to either write something that will allow users to change their passwords, which is more difficult (although not impossible, just watch file locking and permission issues) than using a DB. The option d? Anyway, you can implement that logic yourself in the dialplan, I believe. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + XEN does it make sense?
Jean-Michel Hiver wrote: Hi List, I was wondering if anybody had tried running Asterisk inside virtualization software such as Xen. Are there known problems doing it? We run a number of systems with Xen, its great once you figured out the nags of it :) Remember, to do anything with hardware you will still need Xen 2, not Xen 3. Florian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] simulating a few thousand SIP clients?
Roy, Wai Wu wrote: sure, but I need to simulate the SIP REGISTER and OPTION traffic sent by ATAs as well. What is the current registration time you accept on the servers ? 3600 ?? One thing you can do to try this is set a number of devices to a much shorter registration period. This will effectively deliver just as many REGISTER commands so it can be used for a reasonable test. We've used 10 phones at a registration time of 1 second to 'emulate' 1200 phones at a registration time of 120 seconds. This will ofcourse not emulate the call volume, only the REGISTRERs (and perhaps OPTIONs). Florian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Access Codes
Ronald Wiplinger wrote: Dakota wrote: I would like to setup Asterisk as follows: When users make inter-office calls they can dial the extensions, however if they want to make an external call, that they enter a code on their phone, before they external call can go through. We would like to give each user an access code, this way we can limit certain employees from making certain calls to certain places. What's the best recommendation for this? Use astcc (prepaid card system) with pin! You can give the rates all to free if you want. Give each one who is allowed a balance, if they do not have, they cannot call. You also would have a nice statistic for each call, bye Ronald Wiplinger Nice one Ronald. Keep in mind though that ASTCC has an issue with free calls. There's a workaround in the wiki. Not a show stopper though, even for what he wants. Why not use Authenticate()? You can then set each person to be under a certain context based on password or extension. Under each person's context, include other contexts that allow/limit their capability to call. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
