[asterisk-users] Astmanproxy Yahoo Group
Hi All, Is the astmanproxy yahoo group (http://tech.groups.yahoo.com/group/asterisk-astmanproxy/) still working? It seems to me the most recent posts are 2006's. I have sent a message but didn't receive any feedback and the post was not listed. Is the project still being maintained? Is there any other discussion forum about? Thanks for any update Best regards___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Listening to agent's conversation while waiting in the queue
Hello, I would like to configure a queue with only one agent. But while waiting the callers in the queue should listen to the agent's conversation instead of a music on hold. Is that possible with Asterisk? Does someone out there ever work on a similar configuration? Thank you for any suggestion Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hang up as soon as other party picks up call
Hello,
I have an Asterisk box with a TE410P connected to a PRI line and agents with
X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I
make outbound calls it hangs up as soon as other party tries to picks up the
call. Does someone ever experienced this situation? On X-Lite, only
G711-ulaw is enabled and here is what i put in sip.conf:
[4001]
type=friend
username=4001
secret=4001
host=dynamic
context=callout
disallow=all
allow=ulaw
And below is what i get from Asterisk debug.
Aug 2 11:04:17 DEBUG[180235]: chan_sip.c:5320 check_user_full: Setting NAT
on RTP to 0
Aug 2 11:04:17 DEBUG[180235]: chan_sip.c:825 __sip_ack: Stopping
retransmission on '[EMAIL PROTECTED]' of
Response 37605: Found
Aug 2 11:04:17 DEBUG[180235]: chan_sip.c:5320 check_user_full: Setting NAT
on RTP to 0
Aug 2 11:04:17 DEBUG[180235]: chan_sip.c:7140 handle_request: Check for res
for 4001
Aug 2 11:04:17 DEBUG[180235]: chan_sip.c:1652 update_user_counter: Call
from user '4001' is 1 out of 0
Aug 2 11:04:17 DEBUG[180235]: chan_sip.c:4538 build_route: build_route:
Contact hop: sip:[EMAIL PROTECTED]:5060
-- Executing Dial(SIP/4001-40ee, Zap/g1/6152437) in new stack
-- Called g1/6152437
Aug 2 11:04:17 DEBUG[557083]: rtp.c:1166 ast_rtp_write: Ooh, format changed
from UNKN to ULAW
Aug 2 11:04:22 DEBUG[262160]: chan_zap.c:1186 zt_enable_ec: Enabled echo
cancellation on channel 1
-- Zap/1-1 is ringing
Aug 2 11:04:22 DEBUG[557083]: channel.c:1436 ast_indicate: Driver for
channel 'SIP/4001-40ee' does not support indication 3, emulating it
Aug 2 11:04:22 DEBUG[557083]: channel.c:1551 ast_prod: Prodding channel
'SIP/4001-40ee'
Aug 2 11:04:22 DEBUG[557083]: channel.c:1128 ast_settimeout: Scheduling
timer at 160 sample intervals
Aug 2 11:04:22 DEBUG[557083]: channel.c:1128 ast_settimeout: Scheduling
timer at 0 sample intervals
-- Channel 0/1, span 1 got hangup
Aug 2 11:04:28 DEBUG[557083]: chan_zap.c:2427 zt_setoption: Set option
AUDIO MODE, value: ON(1) on Zap/1-1
Aug 2 11:04:28 DEBUG[557083]: chan_zap.c:1940 zt_hangup: Hangup: channel: 1
index = 0, normal = 28, callwait = -1, thirdcall = -1
Aug 2 11:04:28 DEBUG[557083]: chan_zap.c:2076 zt_hangup: Not yet hungup...
Calling hangup once with icause, and clearing call
Aug 2 11:04:28 DEBUG[557083]: chan_zap.c:1218 zt_disable_ec: disabled echo
cancellation on channel 1
Aug 2 11:04:28 DEBUG[557083]: chan_zap.c:2339 zt_setoption: Set option TDD
MODE, value: OFF(0) on Zap/1-1
Aug 2 11:04:28 DEBUG[557083]: chan_zap.c:1161 update_conf: Updated
conferencing on 1, with 0 conference users
Aug 2 11:04:28 DEBUG[557083]: chan_zap.c:2421 zt_setoption: Set option
AUDIO MODE, value: OFF(0) on Zap/1-1
Aug 2 11:04:28 DEBUG[557083]: chan_zap.c:1218 zt_disable_ec: disabled echo
cancellation on channel 1
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
Aug 2 11:04:28 DEBUG[557083]: app_dial.c:1025 dial_exec: Exiting with
DIALSTATUS=NOANSWER.
-- Registered '2002' (AUTHENTICATED) at 192.168.1.41:4569
-- Registered '3002' (AUTHENTICATED) at 192.168.1.54:4569
Aug 2 11:04:38 WARNING[557083]: pbx.c:1933 ast_pbx_run: Timeout, but no
rule 't' in context 'callout'
-- Executing SetCDRUserField(SIP/4001-40ee, STATUS=NOANSWER) in new
stack
Aug 2 11:04:38 DEBUG[557083]: cdr_addon_mysql.c:178 mysql_log: cdr_mysql:
inserting a CDR record.
Aug 2 11:04:38 DEBUG[557083]: cdr_addon_mysql.c:197 mysql_log: cdr_mysql:
SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration
,billsec,disposition,amaflags,accountcode,userfield) VALUES ('2005-08-02
11:04:17','\4001\ 4001','4001','6152437','callout',
'SIP/4001-40ee','Zap/1-1','SetCDRUserField','STATUS=NOANSWER',21,0,'NO
ANSWER',3,'','STATUS=NOANSWER')
Thanks for any tips
Lamine
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[Asterisk-Users] Prad or V5.2
Hello, Does someone out there ever heard about PRAD or V5.2. Is there any link with Digium's TE110P? Thanks for any enlightments Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to decrease Asterisk load
Should I believe that at this time there is no DSP capable cards working with Asterisk? - Original Message - i From: izo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 13, 2005 6:11 AM Subject: Re: [Asterisk-Users] How to decrease Asterisk load On 5/12/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Digium cards do not have a built in DSP. Neither do the Sangoma as far as I know. I don't know about VoiceTronix. As yet ! As for digium cards latest cvs commits suggest that there is some ongoing development on hardware based echo cancelation and dtmf detection. So its just a matter of time. rgrds m. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IPVolution release info....
Thanks for this precision !! Certainly, a good news for Asterisk users community. - Original Message - From: Wiley Siler To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, May 12, 2005 10:16 PM Subject: [Asterisk-Users] IPVolution release info From atacomm From: Jessee J Holmes [mailto:[EMAIL PROTECTED] Sent: Thursday, May 12, 2005 2:24 PMTo: Wiley SilerSubject: Re: Got a date yet? No specific release date as of yet; but, we're hoping to have a physical date soon. So far planned release is either in June or July. Right now they developers are cleaning up the echo cancellation code on the chip andfinalizing things in that aspect. So far everything has been looking good and positive. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ Atacomm can also provide you with competitive rates from your local carriers. Remember: E-mail is not a secure medium. Please do not send payment information via e-mail. On May 12, 2005, at 1:27 PM, Wiley Siler wrote: ipVolution TDM60 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to decrease Asterisk load
Hi everybody, I would like to decrease the load of my asterisk server. Could someone recommend me a solution? I have thought about a hardware component that would do some tasks as compression/decompression or codec translations but wonder if such a solution exist. Thanks for any suggestion Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to decrease Asterisk load
Thanks Mike, I am already using rawplayer for music-on-hold. I have been told of IpVolution TDM60 card that has DSP resources ... Does someone out there ever experienced it? Lamine - Original Message - From: Mike Holloway [EMAIL PROTECTED] To: Mamadou Lamine KA [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 12, 2005 6:20 PM Subject: Re: [Asterisk-Users] How to decrease Asterisk load One thing I do is use rawplayer instead of mpg123 for music-on-hold playback, so that mp3's don't have to be decompressed in realtime. See the wiki for details on using sox to convert your audio samples to raw format, and how to configure musiconhold.conf to use rawplayer to play these files. -mike Mamadou Lamine KA wrote: Hi everybody, I would like to decrease the load of my asterisk server. Could someone recommend me a solution? I have thought about a hardware component that would do some tasks as compression/decompression or codec translations but wonder if such a solution exist. Thanks for any suggestion Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Quality
Hello David, Bad voice quality may be caused by many reasons. I suggest you test the two servers separately first. Monitor CPU load during calls in each server and verify if the communication devices used by asterisk (voice boards, network interfaces ... ) don't share interruptions. In iax.conf try to replace the two lines jitterbuffer=200 jitterbuffer=yes with the six following jitterbuffer=yes dropcount=2 maxjitterbuffer=500 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1 You may also want to take a look at http://www.voip-info.org/wiki-QoS Regards Lamine - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 2:51 PM Subject: [Asterisk-Users] Voice Quality Hello, I have setup two * servers and they are communicating using IAX. I'm passing calls from SRV A (internet connection T1) to SRV B (internet connection: 512). For some reasons I have an issue with the quality. The voice is a bit scratchy. I have tried iLBC and SPEEX, but it didn't make any difference. Now, assuming that I have an issue with Bandwidth, what would be the best way to configure my iax.conf. (A bit confused about jitterbuffer and tos) Here is my iax.conf @ location A: [general] port=4569 bandwidth=low disallow=all allow=ilbc ;allow=ulaw ;allow=speex jitterbuffer=200 jitterbuffer=yes tos=lowdelay and iax.conf @ location B: [general] port=4569 bandwidth=low disallow=all allow=ilbc ;allow=ulaw ;allow=speex jitterbuffer=200 jitterbuffer=yes tos=lowdelay [guest] type=user context=default callerid=Guest IAX User disallow=all allow=ilbc Thanks guys ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ACD in Asterisk
Hi everybody, I am having a problem while setting up queues in Asterisk. Callers are kept in the queues and told to wait while there are available agents. Even if I use ringall as strategy the call is not always sent to all free agents. Is there a problem with Automatic Call Distribution in Asterisk or am I missing something? Below is my queues.conf. Thanks for any suggestion Lamine [general] [default] [sceclient] music=sceclient strategy=leastrecent timeout=30 retry=5 wrapuptime=0 maxlen=0 announce-holdtime=no member=Agent/3001 member=IAX2/3000 member=IAX2/3001 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] joinempty=no
Hello, I would like to know if there is a simple way to redirect callers to another extension (may be an IVR) when no agent is logged on. joinempty is set to no in my queue. Thank you for any tip Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chanspy and current version of cvs
Hi everybody ! I had patched asterisk to install chanspy weeks ago and everything was ok. With the current version of cvs i am having failures when i try to apply the same patch and the url where i originally downloaded it seems no longer active. Is the patch any longer maintained or has it been replaced with another function. Thanks in advance Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call asterisk from perl
You can also use the manager. Take a look at http://www.voip-info.org/wiki-Asterisk+manager+API - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 6:12 AM Subject: Re: [Asterisk-Users] Call asterisk from perl Ousmane Doukara wrote: Is it possible to call asterisk from a script ? I have a script scheduled in cron and I want to be able to Dial a number from that script whenever an event occur. Look on the wiki for outgoing spool files. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDMO4B, GSM Gateways and CallerID
Hello everybody, I have an Asterisk box with a TDM04B and would like to connect it to a GSM Gateway. Can someone tell me whether i can get the callerid for incoming calls in this case? Thanx Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agents question
Hi, Is there a way to know whether a logged agent is in communication or not? I would like the supervisor to select the agent he wants to spy. Thanks for any suggestion Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ChanSpy Usage
Hi, Could someone tell me the significance of arguments in chanspy synopsis: Chanspy([-opts|]chan_name|scan[|scanspec]). What are the possible values for opts,scan and scanspec? Thanx Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Signaling / Streaming
Hi, With Gnugk, make sure the proxy mode is not enabled if you want voice to pass directly from endpoints. Regards Lamine - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:21 AM Subject: [Asterisk-Users] Signaling / Streaming Hi When I forward calls from SER (or GNUGK) to Asterisk, the SER ( or GNUGK) are just used for signaling, but the call streaming passes from the endpoint directly to Asterisk, isnt it? Or does the streming passes from the Endpoint to SER and then to the Asterisk? Thanks Joao Pereira ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Signaling / Streaming
Yes, This mode is generally used when some endpoints have private addresses behind a NAT while others have public addresses. In this case all the traffic passes through the GK. Take a look at paragraph related to Proxy at http://www.gnugk.org/gnugk-manual-4.html#ss4.2 Lamine - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 11:18 AM Subject: Re: [Asterisk-Users] Signaling / Streaming Ok, then I guess the way we use SER and GNUGK to redirect calls to Asterisk makes the diference. If we are using them as proxy, the stream will pass through them, if we dont use proxy, they will be used just for signaling. Joao - Original Message - From: Mamadou Lamine KA [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:50 AM Subject: Re: [Asterisk-Users] Signaling / Streaming Hi, With Gnugk, make sure the proxy mode is not enabled if you want voice to pass directly from endpoints. Regards Lamine - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:21 AM Subject: [Asterisk-Users] Signaling / Streaming Hi When I forward calls from SER (or GNUGK) to Asterisk, the SER ( or GNUGK) are just used for signaling, but the call streaming passes from the endpoint directly to Asterisk, isnt it? Or does the streming passes from the Endpoint to SER and then to the Asterisk? Thanks Joao Pereira ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitoring
What version of sox do you use? Lamine - Original Message - From: Robert Spielmann [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 2:40 PM Subject: [Asterisk-Users] Monitoring Hi, I have some trouble with the Monitor() application. I start and stop it via the management interface, giving no special parameters except the channel name. What happens is: - if I specify WAV as the format, the resulting files are exactly 44 bytes big and contain nothing at all - if I specify GSM as the format, the resulting files are of size 0. I did not request mixing of the files or anything else. Any ideas why the monitoring fails? Cheers Robert Spielmann - TAL.DE Klaus Internet Service GmbH [EMAIL PROTECTED] Robertstr. 6 * D-42107 Wuppertal, Germany Tel +49 (0) 202 495-364 * Fax +49 (0) 202 / 495-399 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 problem
As you have a TDM400 you should load wcfxs module ( modprobe wcfxs or modprobe wctdm for new versions) regardless to modules you have installed on your board. You should also check the signalling specified in zaptel.conf according to your modules and the order they are placed on your TDM400. Best regards Lamine - Original Message - From: Steven P. Donegan [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, December 27, 2004 6:01 PM Subject: [Asterisk-Users] TDM400 problem I recently swapped 2 FXO modules on to what had previously been a 4 FXS version of the TDM400 board. The FXS ports are recognized - the FXO ports don't appear to be recognized (ie modprobe wcfxo and ztcfg both say channel 1 isn't there). Has anyone experienced this problem? All software is current as of this AM. If the old FXS modules are re-installed all works just dandy (other than the fact that I need the 2 FXO ports)... Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] where I can find some learning book about asterisk?
Hello, Take a look at http://www.signate.com You can also find various documentation resources at http://www.voip-info.org/tiki-index.php?page=Asterisk Regards Lamine - Original Message - From: FCG ZHAO Zigang [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, December 24, 2004 2:05 AM Subject: [Asterisk-Users] where I can find some learning book about asterisk? Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -- : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] : 20041224 7:51 : asterisk-users@lists.digium.com : Asterisk-Users Digest, Vol 5, Issue 350 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: rtp channels not through asterisk (Brian West) 2. Turning * Hangup off in queues ([EMAIL PROTECTED]) 3. Re: Voicemail email notification (Rich Adamson) 4. Can't Make Outgoing Call (Norman Zhang) 5. Re: Voicemail email notification (Dorn Hetzel) 6. Re: Asterisk in parallel with PSTN [OT] (Rich Adamson) 7. Re: rtp channels not through asterisk (Rich Adamson) 8. Re: Realtime sipbuddies table structure why? (Greg - Cirelle Enterprises) 9. RE: Polycom Buddies (Paul Hales) 10. Re: Queue - roundrobin member order (Adam Goryachev) 11. Re: Voicemail email notification (Rich Adamson) 12. Re: Can't Make Outgoing Call (Norman Zhang) 13. Re: Recommended IAX softphone. (Bruno Hertz) 14. Re: sip seeding vs registration (Greg - Cirelle Enterprises) 15. Asterisk 1.0.3 no RedHat zaptel script? (Jerry Geis) 16. Re: Recommended IAX softphone. (Erik Espinoza) -- Message: 1 Date: Thu, 23 Dec 2004 16:51:22 -0600 From: Brian West [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] rtp channels not through asterisk To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII canreinvite=yes Aterisk stays in the signaling path so unless you're running tcpdump or the like you'll never notice this. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of bijan Sent: Thursday, December 23, 2004 4:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] rtp channels not through asterisk In wiki pages it is stated that The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge. Currently with my settings, I notice that all rtp's are passing through my asterisk. How could I achieve that they go directly from phone to phone? I assume this way, my machine will have less load and therefore could handle more calls. regards Bijan Karimi -- Message: 2 Date: Thu, 23 Dec 2004 19:16:19 -0600 (CST) From: [EMAIL PROTECTED] Subject: [Asterisk-Users] Turning * Hangup off in queues To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII Hi ! Can somebody tell me how to turn the * Hangup option utrned off in queues. I have not used any H option but still as an agent if I press * key the user gets disconnected. Somehow it is turned on by default. Can I turn this option off In my extensions.conf I have written : exten = 8000,3,Queue(supportq|t) plz help me inthis regard ... Thanks ! Usman. -- Message: 3 Date: Thu, 23 Dec 2004 16:51:34 -0600 From: Rich Adamson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Voicemail email notification To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 Are there any common silent failure modes for email notification from the Voicemail module. I put the email and pager email addresses in my entry in voicemail.conf but no mail gets sent when I leave a voicemail. No obvious error messages either, unless I'm just not looking in the right place. Thanks for any clues :) Nop, that's it other then you have to have sendmail configured and running on the system (or have a substitute mail handler). Rich -- Message: 4 Date: Thu, 23 Dec 2004 14:58:04 -0800 From: Norman Zhang [EMAIL PROTECTED] Subject: [Asterisk-Users] Can't Make Outgoing Call To: Asterisk Users Mailing List - Non-Commercial Discussion
Re: [Asterisk-Users] asterisk at large
Hello *'s, First Of all Marry Christmas, I want to setup asterisk at large means my main asterisk server placed in my office(in Pakistan), and some offices outside Pakistan and i want to connect these locations to my main * server (in Pakistan) on remote locations i'll used asterisk can i do this or may be i changed my plans Yes you can. Register your remote servers to your main server and choose different numbers for different Asterisk servers. Detailed informations are available at http://www.voip-info.org/wiki-Asterisk+-+dual+servers Regards Lamine kindly guides me. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with sox
Hello, I have installed sox-12.17.6 from sources with 2.4.27 kernel but i can't mix audio files. (For both wav and gsm formats) For example, when i try soxmix filename-in.wav filename-out.wav filename.wav. Everything seems ok. There no error messages. But the output file (filename.wav) is empty, as if the volume was turned off. Am I doing something wrong or is that a bug? Thanks for any hint Lamine ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users]Re: Problem with sox
Thanks Dawson,
12.17.5 version works fine but i have got to mix files in gsm format first
before converting it to wav by using sox. But soxmix doesn't work directly
with wav files.
Altus, try using groups in zapata.conf. This can be done by adding
group=1
callgroup=1
pickupgroup=1
before channels vpb/1-3 and vpb/1-4
and then in extensions.conf you will have something like
exten = _0.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = _0.,2,Monitor(wav,${CALLFILENAME},m)
exten = _0.,3,Dial(vpb/g1/${EXTEN:1})
exten = _0.,104,Congestion
Hope this help
Lamine
I installed the new version of asterisk
But the other probelm I got was,were are using the voicetronix cards,so
if you go and put
ignorepat = 0
exten = _0.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = _0.,2,Monitor(wav,${CALLFILENAME},m)
exten = _0.,3,Dial(vpb/1-3/${EXTEN:1})
exten = _0.,4,Dial(vpb/1-4/${EXTEN:1})
exten = _0.,5,Congestion
exten = _0.,104,Congestion
And make a call it will go out on vpb/1-3
But when another call is maed it will keep on trying to go out on
vpb/1-3 instead of detecting a busey signal and moving to vpb/1-4
If I take out the whole moniter part it works 100?
Beats me
Altus
dawson wrote:
It's appears to be broken in that version.
Go back to sox-12-17.5(it should work).
Hello,
I have installed sox-12.17.6 from sources with 2.4.27 kernel but i can't
mix
audio files. (For both wav and gsm formats)
For example, when i try soxmix filename-in.wav filename-out.wav
filename.wav. Everything seems ok. There no error messages. But the
output
file (filename.wav) is empty, as if the volume was turned off. Am I doing
something wrong or is that a bug?
Thanks for any hint
Lamine
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Re:[Asterisk-Users] Setting jitterbuffer in with iax
Hello everybody; I would like to know the parameters on which depend jitterbuffer in iax.conf. Is there some kind of formula to set the correct values? Thanks in advance for any help Lamine I'd say that the numbers in the iax.conf.sample are a good balance. You'll also find quite a lengthy explanation of the fields in that sample file. Regards, Steve Very nice! I have set the values and I am having a far better quality. Thanks Lamine ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting jitterbuffer in with iax
Hello everybody; I would like to know the parameters on which depend jitterbuffer in iax.conf. Is there some kind of formula to set the correct values? Thanks in advance for any help Lamine ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VICIDIAL and IAX
Hello everybody, I would like to know if there is a support of IAX in vicidial. I want to make predictive dialing use vicidial using IAX soft phones. Thanks in advance Lamine ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZapBarge and SIP Channels
Hello everybody, Is there any alternative to Asterisk ZapBarge command for SIP and IAX channels? Thanks Lamine
[Asterisk-Users] fax detection and X100P
i have successfully updated my cvs pull of zaptel but for asterisk when i type "make clean"i have the folowing error: Makefile:73: *** missing separator. Arrêt ( Arrêt means stop) Lamine
[Asterisk-Users] fax detection and X100P
Hi everybody I am having problem detecting fax with my X100P. I have RedHat 8 as OS and an X100P and a TDM400P. The X100P being plugged into PSTN. I have successfully installed tiff-v3.5.7 and spandsp-0.0.1 and also patched Asterisk wthout problem. Here is my zapata.conf file context=cda signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived faxdetect=both channel = 1 ; ;TDM400P ; context=cda signalling=fxo_ks callwaiting=yes callwaitingcallerid=yes echocancel=yes echocancelwhenbridged=yes threewaycalling=yes transfer=yes callerid=MSY 103 mailbox=103 channel = 2 and here is my extension.conf file [cda] exten =s,1,Answer ;Operators exten = 110,1,SetLanguage(fr) exten = 110,2,AgentLogin exten = 15,1,SetLanguage(fr) exten = 15,2,VoicemailMain ;Clients campagne natural mystic exten = 12345,1,SetLanguage(fr) exten = 12345,2,Answer exten = 12345,3,Queue(110) ; Pour recevoir les faxes exten =fax,1,RxFax(/var/spool/asterisk/incoming/lamine.tif) ; Pour envoyer le fax exten =8236331,1,TxFax(/var/spool/asterisk/incoming/lamine.tif) But when I launch Asterisk i get the following message: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Jul 5 16:22:13 WARNING[16384]: chan_zap.c:7890 setup_zap: Ignoring faxdetect and there is a timeout when i try to receive fax. I have checked in dsp.c and the line #define FAX_DETECT is not commented Has someone ever encountered this problem? Any help would be greatly appreciated Lamine ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting Events in queues
Hi everybody, I would like to know how can I detect events in queues. For example when an operator answers a call which was in a queue I would like to some informations related to the caller (quote from the database) to the operator. I saw these events in queuelog.conf but i don't know how to retrieve them in my AGI. Thanks for any help Lamine ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Dev] SPAM MESSAGE - [Asterisk-Users] warning message (sound card) - when I run asterisk!!!
Hi Neo,
Your sound card is not well configured. Try to find
the right driver and load the correct module for it. ALSA (www.alsa-project.org/)
may help for this.
Hope this can help
Lamine
- Original Message -
From:
Neo Jia
To: [EMAIL PROTECTED] ; [EMAIL PROTECTED]
Sent: Wednesday, May 26, 2004 7:06
PM
Subject: [Asterisk-Dev] SPAM MESSAGE -
[Asterisk-Users] warning message (sound card) - when I run asterisk!!!
All,
After installing asterisk on Linux, I run "asterisk
-vvvc". But I got the following warning message:
chan_oss.so] = (OSS Console Channel Driver)
May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980
load_module: XXX I don't work right with non-full
duplex sound cards XXX
== Registered channel type 'Console' (OSS Console
Channel Driver)
== Parsing '/etc/asterisk/oss.conf': Found
May 26 00:37:58 WARNING[-1168819280]: chan_oss.c:238
sound_thread: Read error on sound device: Resource
temporarily unavailable
[chan_phone.so] = (Linux Telephony API Support)
== Parsing '/etc/asterisk/phone.conf': Found
== Registered channel type 'Phone' (Standard Linux
Telephony API Driver)
My sound card information:
Vendor : Intel Corp.
Model : 82801CA/CAM AC'97 Audio Controller
Module : i810_audio
After running 'dial' command under the asterisk
prompt, I got the following message without any sound.
*CLI -- Executing Wait("OSS/dsp", "1") in new
stack
-- Executing Answer("OSS/dsp", "") in new stack
Console call has been answered
-- Executing DigitTimeout("OSS/dsp", "5") in new
stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout("OSS/dsp", "10") in
new stack
-- Set Response Timeout to 10
-- Executing BackGround("OSS/dsp",
"demo-congrats") in new stack
May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:408
soundcard_setinput: Unable to re-open DSP device:
Device or resource busy
May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:567
oss_write: Unable to set device to input mode
May 26 00:40:55 WARNING[-1221268560]: file.c:537
ast_readaudio_callback: Failed to write frame
-- Playing 'demo-congrats' (language 'en')
== Spawn extension (local, s, 5) exited non-zero on
'OSS/dsp'
Is there anyone can give me any hints or help?
Thanks,
Neo
[Asterisk-Users] Call recording between SIP phones
Hi everybody, I have been searching around for days on how to record calls between SIP phones.Could someone tell me whether it is possible? The Record command doesn't seem to work during a call. Thanks Lamine ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial and MeetMe on the same channel
Hello everybody, I would like to know whether it is possible to run Dial and MeetMe commands simultaneoously on the same channel. I am using a C AGI as below but it seems to me that only the first command that is called in the agi is executed. ... // Préparation de la commande pour l'appel du client fprintf(stderr,%s%s,numtocall, is the number to call\n); strcpy(cmd,EXEC Dial ); strcat(cmd,numtocall); //numtocall is a variable quote from teh database strcat(cmd, 60); // Exécution de la commande et libération du buffer fprintf(stderr,%s\n,cmd); printf(%s\n,cmd); fflush(stdout); resultcode = checkresult(); // Mise en conférence de l'operateur strcpy(cmd1,); strcpy(cmd1,EXEC MeetMe ); strcat(cmd1,confroom); //confroom is a variable quote from teh database strcat(cmd1,|q); fprintf(stderr,%s\n,cmd1); printf(%s\n,cmd1); fflush(stdout); .. Any reason on why only the first command is successfull?? Thanks in adavance. Lamine ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie Start Question
Hi Everybody, I am very new to Asterisk. I want to set up a PBX and an IVR server with it. I have a wildcard X100P and a TDM400P on my RedHat box. I have installed Asterisk and the devices and everything seems OK. (Asterisk Ready) Now I want to launch the Demo context in /etc/asterisk/extensions.conf so that when a call comes it is directed on that context. How shall I proceed? I have of course read the Asteriskhandbook but it is too the theorical to me. Could someone tell me where i can find exact informations on how to set up and how to use IVR server with Asterisk. Any help will be highly appreciated. Thanks in advance Mamadou Lamine KA
