[asterisk-users] How to associate agents - extensions?
Hi! I am setting up a little call center, but don't know how the agents system works, can you guys please give me a little help? I need to know how asterisk will know when I log agent X, and asterisk know that agent is in the IP Z with the extension Y. Thanks a lot. Hugs, ARPE -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk log format
Hi mates! Please, I need to understand how to search for an specific log by date/time on asterisk logs, but can't understand how this works, can you guys please give me an example about how those logs works? Best regards, Asterisk Guy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1and1 dedicated servers have been down for a few hours .
1and1 dedicated server's service has been down for a few hours , unable to reach them by phone or email. do anyone know what is going on there ? Mario ___ --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
Re: [asterisk-users] Wake-Up Call didn't work
1 there is a correct file in /var/spool/asterisk/outgoing 2 i run asterisk -r to monitor it , it gives out the following error -- Attempting call on Local/[EMAIL PROTECTED] for application MusicOnHold() (Retry 1) Jul 24 08:23:17 NOTICE[21177]: chan_local.c:479 local_alloc: No such extension/context [EMAIL PROTECTED] creating local channel Jul 24 08:23:17 NOTICE[21177]: channel.c:2409 __ast_request_and_dial: Unable to request channel Local/[EMAIL PROTECTED] ( but i have a extension 6009 login to * ) , what is the problem? On 7/23/07, James FitzGibbon [EMAIL PROTECTED] wrote: On 7/23/07, Dovid B [EMAIL PROTECTED] wrote: Can it be that asterisk does not have permission to copy the file over ? Also check your date settings on the server. Yes, it's interesting that the page intro includes the sentence Lots of error checking to make sure its done correctly, but the final step that makes the process work (ensuring that the callfile ends up in the directory that pbx_spool is watching) doesn't have any error checking: touch( $wakefile, $time_wakeup, $time_wakeup ); rename( $wakefile, $callfile ); The fact that you see files in /tmp when all is said and done means that at least some of the script is working. A few things to check: Do the files in /tmp have the correct timestamp (file matches the requested wakeup time)? If so, then everything preceeding the rename seems to have worked, so check if the user running the AGI can move files from /tmp to /var/spool/asterisk/outgoing. Though given that it's an AGI being run by *, you'd have to have a pretty strange setup for that to fail. Perhaps the outgoing directory just doesn't exist (was never created for some reason?) If the files don't have the correct timestamp, start following the logic backwards. Do they look complete? Look through the AGI for places where the wakeup file is written to (i.e. fputs( $wuc, maxretries: $parm_maxretries\n); ) and check that everything that should be written is being written Working backwards you should be able to figure out where the script is failing, then you can check everything that comes afterwards as the user running the AGI to make sure that permissions and directories are set up properly. -- j. ___ --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 ___ --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] Wake-Up Call didn't work
I have setup wake up call in * ( 1.2crc1) following those instructions http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP i can enter the time after dialing 77 , and i see there is wakeup files in /tmp but * nevers make the wakeup call when it is due , what can be the problem ? what shall i check? Mario ___ --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] sip softphone for PDA window mobile 2003 / 5.0 ?
are there any good softphone on PDA window mobile 2003 / 5.0 ? tried sjphone, sound quality is unacceptable. Mario ___ --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] DID providers in Toronto
hi Can anyone recommend a good DID provider offering numbers in Toronto ? ( 1 very stable 2 support porting numbers from Bell, primus, telus.. ) Mario ___ --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] Softphone for smartphone such as Nokia N90 / 93 / N95
looking for good sip softphone for wifi and 3G network. 1 are there any sip softphone ( with gsm/g723/G729 codec ) for smartphone such as Nokia N90 / 93 / N95 ? 2 are there any sip softphone ( with gsm/g723/G729 codec ) for Window mobile5 Or wm2003 ? 3 How is the sound quality of GSM /G723/729 codec on wifi/3G network? which codec is better for wift/3G ? Jackie ___ --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] open source sip softphone (Window OS version )
sjphone firefly (3rd party version) --source is available ? where? On 6/15/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Jun 15, 2006 at 06:06:40AM -0700, Derek Whitten wrote: Asterisk guy wrote: are there any open source sip softphone (Window OS version )? ___ --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 xlite You're the second one to mention xlite. You mind telling me where I can get its source? sjphone firefly (3rd party version) And even when the source code is availble, it does not make the software open source (at least not in the sence of the definition from http://opensource.org , which seems to be what the OP intended). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.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 ___ --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] open source sip softphone (Window OS version )
are there any open source sip softphone (Window OS version )? ___ --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] Free IAX login
[guest] type=friend context=default isecure=very -it doesn;t work , asterisk shows: Feb 9 08:41:13 NOTICE[29683]: chan_iax2.c:6782 socket_read: Rejected connect attempt from 89.*.8... for the incoming call On 2/7/06, Mark Phillips [EMAIL PROTECTED] wrote: Try adding insecure=very to the guest user account in iax.conf. This should not do a user/pass challenge on the incoming call. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com kevin ling wrote: Not sure answer your question? Try to write some html code and let user register the username password online. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy Sent: Tuesday, February 07, 2006 7:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Free IAX login how to set up iax.conf , so IAX clients with any user name and any secret can login to * ? ( no authorize for login ) ___ --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] Free IAX login
[guest] type=friend context=default insecure=very -it doesn;t work , asterisk shows: Feb 9 08:41:13 NOTICE[29683]: chan_iax2.c:6782 socket_read: Rejected connect attempt from 89.*.8... for the incoming call On 2/7/06, Mark Phillips [EMAIL PROTECTED] wrote: Try adding insecure=very to the guest user account in iax.conf. This should not do a user/pass challenge on the incoming call. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com kevin ling wrote: Not sure answer your question? Try to write some html code and let user register the username password online. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy Sent: Tuesday, February 07, 2006 7:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Free IAX login how to set up iax.conf , so IAX clients with any user name and any secret can login to * ? ( no authorize for login ) ___ --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] Free IAX login
for sip.conf , there is a configure option for this : allowguest=yes is there a silimiar setting for IAX ? On 2/7/06, Mark Phillips [EMAIL PROTECTED] wrote: Try adding insecure=very to the guest user account in iax.conf. This should not do a user/pass challenge on the incoming call. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com kevin ling wrote: Not sure answer your question? Try to write some html code and let user register the username password online. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy Sent: Tuesday, February 07, 2006 7:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Free IAX login how to set up iax.conf , so IAX clients with any user name and any secret can login to * ? ( no authorize for login ) ___ --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] Free IAX login
how to set up iax.conf , so IAX clients with any user name and any secret can login to * ? ( no authorize for login ) ___ --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 adjust volume
how to adjust voice volume for sipura 2000 and cisco ata186? ___ --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 1.2 Released!
does it include the patch for VAD? ( dropping extra frame of G.729 since we already have a VAD frame at the end ) Mario ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g.729 pass thru mode
How to set asterisk in pass-through mode ? could you give a sample configure for passthrough? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 Released!
does the following patch work for 1.2? how to apply it to 1.2? ( I am not a programmer, don't know how to use .diff file). http://bugs.digium.com/view.php?id=5374 silence-suppression-2.diff On 11/17/05, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Asterisk guy wrote: does it include the patch for VAD? ( dropping extra frame of G.729 since we already have a VAD frame at the end ) It does not include several important things. It does not include a SIP jitter buffer. It does not include the ability to use Zaptel for timing of the RTP audio. It does not include VAD/CND support. As far as I know it also does not have the patch to make the new IAX2 jitterbuffer work correctly when connecting to a 1.0.x server. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dropping extra frame of G.729 since we already have a VAD frame at the end
does 1.20rc2 includes the patch? will this patch be included in 1.20rc3? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum Number of SIP Phones Supported By Asterisk
Is the asterisk server actually pushing the bits for a call or just doing call setup and connecting the two endpoints directly? how to force asterisk just doing call setup and connecting the two endpoints directly?reinvite=yes ? if UA is behind NAT with reinvite=yes , will asterisk actually push the bits or drop the call? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dropping extra frame of G.729 since we already have a VAD frame at the end
dropping extra frame of G.729 since we already have a VAD frame at the end- just tested 1.20rc1, it is still there. where to get patch for 1.20rc1? does 1.20rc2 includes the patch? On 10/29/05, Kanishka Somaratne [EMAIL PROTECTED] wrote: Hi I get the following error when i make a call from 729 to 729 dropping extra frame of G.729 since we already have a VAD frame at the end I am using asterisk 1.0.9, there was a patch for CVS vertion, is there a patch for 1.0.9 tks kani ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to play a voice file for decline
When get sip respond 6xx ( such as 603 decline), I want asterisk to play a voice file to the caller, how to do this in extensions ? for example, when get 603 respond, play decline.gsm to caller when get 604 respond, play doesnot-exit.gsm to caller when get 606 respond , play not-acceptalbe.gsm to caller ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] play a voice file voice for decline
When get sip respond 6xx ( such as 603 decline), I want asterisk to play a voice file to the caller, how to do this in extensions ? for example, when get 603 respond, play decline.gsm to caller when get 604 respond, play doesnot-exit.gsm to caller when get 606 respond , play not-acceptalbe.gsm to caller SIP response codes, class 6: Global failures 600 Busy Everywhere 603 Decline 604 Does Not Exist Anywhere 606 Not Acceptable ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transcode or passthrough
Are there any command to show a calls is in transcoding or passthrough codec mode ? Mario ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as H323 gateway
I have setup Asterisk with chan_oh323 to connect to the PSTN over H.323 in --may i know which version of asterisk and oh323? On 10/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Juanjo, can you provide some more detail about which version you are using both for asterisk and OpenH323, the hardware dimensioning and the amount of traffic you manage with this solution: how many lines, codecs you use? We should manage a full blown PRI (30 channels), the server is SuperMicro with P4DP8G2 motherboard, dual Xeon 2,8 GHZ, 2 GB RAM and dual SCSI 15K HD in RAID 0. We plan to use G729 codec and a Digium TE110P Card. Any detail will be very useful brgds Francesco Pellegrini ++ | Frame Srl | | Via Antonio Cantore 62/10 | | 16149 Genova | | Tel. +39 010 8680570| | Fax. +39 010 6591413 | | Cell. +348 2237798 | ++ On 10/4/05, Juan Jose Comellas juanjo at comellas.com.ar wrote: I have setup Asterisk with chan_oh323 to connect to the PSTN over H.323 in Buenos Aires, Argentina. Currently I'm using direct connections to the telephone company's (iplan) H.323 gateway, but I'm working on using an intermediate H.323 gatekeeper to take advantage of the telephone company's redundant servers. I think the telco uses Cisco hardware, but I'm not completely sure. We've just started using this, but it seems stable so far. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Oh323 and Caller ID missing
After more test, get the following: Oh323's call-id works on asterisk1.07+ oh323 (0.66), doesn't work on ( asterisk1.2.0beta1+oh323 0.73), any pathch to get oh323 0.73 works? On 10/2/05, Asterisk guy [EMAIL PROTECTED] wrote: I get the same problem. ( asterisk1.2.0beta1+oh323 0.73), any suggestion for this ? On 6/13/05, Federico Alves [EMAIL PROTECTED] wrote: I am sending calls using Oh323 to a Cisco Gateway (AS5300), and although I set the caller id correctly in my perl AGI script $AGI-set_callerid($ani); , the gateway does not see any caller id coming from my Asterisk box. I use the very latest version of Oh323 as published in the Inaccess web site, and Asterisk HEAD from two days ago. The caller ID important for this client, because he will further authenticate the call based on the ANI. I am only doing a codec conversion. Any help is appreciated from Jeremy McNamara. ___ 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as H323 gateway
may i know which version oh323? On 10/4/05, Juan Jose Comellas [EMAIL PROTECTED] wrote: I have setup Asterisk with chan_oh323 to connect to the PSTN over H.323 in Buenos Aires, Argentina. Currently I'm using direct connections to the telephone company's (iplan) H.323 gateway, but I'm working on using an intermediate H.323 gatekeeper to take advantage of the telephone company's redundant servers. I think the telco uses Cisco hardware, but I'm not completely sure. We've just started using this, but it seems stable so far. On Tue October 4 2005 06:28, [EMAIL PROTECTED] wrote: Is there anyone who is currently using Asterisk as a production H323 gateway? And using which combination of asterisk and H323 (chan_h323, chan_oh323?) The main issue is interoperability with other H323 parties (Cisco AS53xx, Nextone, etc). Searching the mailing list it seems that both h323 and oh323 are not so stable, is it only an impression or using h323 is really not so advisable? Francesco Pellegrini [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Oh323 and Caller ID missing
I get the same problem. ( asterisk1.2.0beta1+oh323 0.73), any suggestion for this ? On 6/13/05, Federico Alves [EMAIL PROTECTED] wrote: I am sending calls using Oh323 to a Cisco Gateway (AS5300), and although I set the caller id correctly in my perl AGI script $AGI-set_callerid($ani); , the gateway does not see any caller id coming from my Asterisk box. I use the very latest version of Oh323 as published in the Inaccess web site, and Asterisk HEAD from two days ago. The caller ID important for this client, because he will further authenticate the call based on the ANI. I am only doing a codec conversion. Any help is appreciated from Jeremy McNamara. ___ 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 implementation 0.67 has call-id problem
I am trying oh323(version 0.67) , make call from sip UA to h323 gateway, can't get Call-id pass from sip UA to h323 gateway, h323 always gets call-ID sent from Asterisk as *. are there any configure to pass the correct call-id from sip UA to h323 gateway? or this is a bug in oh323 0.67? how about oh323 0.73 ? Mario On 9/29/05, Kanishka Somaratne [EMAIL PROTECTED] wrote: hi guys I was working on asterisk and h323 for the past 2 weeks i have the following feedback please let me know if i am wrong h323 implementation I managed to install this it works, but the problem is it accecpts all calls from all ips. there is no way i can let it accecpt calls only from the IPs i give and bill depending on IP oh323 implementation managed to install, same as h323 implementation i can't add a list of ips and restrict access, the 729 - 723.1 codec convertion does not work well, get a robort voice ooh323c installeed but do not know how to configure :( woomera let me know if there's any one who has tried this. what i want to do it accecpt h323 calls and bill depending on the ip address and send the calls via h323 depdning on the gateway IP i add Regards Kansihka ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OOH323C
HI I got the following error during compiling ooh323. In function `h323_set_rtp_peer':chan_h323.c:2745: error: structure has no member named `tech_pvt'chan_h323.c: At top level:chan_h323.c:68: error: storage size of `h323_tech' isn't knownmake[2]: *** [chan_h323.lo] Error 1 make[2]: Leaving directory `/root/asterisk-ooh323c-0.2/src'make[1]: *** [all-recursive] Error 1make[1]: Leaving directory `/root/asterisk-ooh323c-0.2'make: *** [all] Error 2 What is the reason ? Mario On 9/30/05, Dan Austin [EMAIL PROTECTED] wrote: Asking which H323 channel is the best turns out to be a deeplypersonal issue, at least noting the responses in the past. I've tried and used all three. Here are my thoughts-Chan_h323 (the original)-Did not work in our environment.Known issues with Cisco'sCall Manager.Other than the requirements for OpenH323 and PWLib, it was easy to setup and configure.Chan_oh323Worked fine for us.Has the same dependencies as chan_h323,also easy to setup and configure.Chan_h323 (ooh323c based)This one has been a winner for us.No dependencies on OpenH323 or PWLib, which while not terrible to build/setup, is extra effortand can be tricky to match known working versions.Setup and configuration has been very simple.If you have configuredthe other channels, this one should seem familiar. A seperate note in favor of the new chan_h323 is the developer support.I found a couple little bugs that related to our use of Cisco CallManager, and expected little or no interest in getting them resolved. I had a test version made available to me in just over a day andcomplete resolution a few hours later.Dan-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of KanishkaSomaratneSent: Thursday, September 29, 2005 7:28 AM To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] OOH323Chihas any one used OOH323C i tried this it is installed but do not knowhow to configure has any one used this, what is the best h323 addon to use withasterisk___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec conversion
for sip calls, asterisk is able to convert a incoming g729 cal to a outgoing G.711 call. Foroh323, I am unable toget asterisk to convert a incoming g729 call to a outgoing G711 call . my question is :For h323, how to configure asterisk to convert a incoming h323/g729 calls to a outgoing h323 g.711 call ? any suggest are welcome. On 1/17/05, Helder Rogério [MICROREDE] [EMAIL PROTECTED] wrote: Hi! Is there any way to receive in * server a call from a Terminal adapter in G.723/ G.729 and then convert it to G.711? I'm wondering this because I can only place all thru Broadvoice in G.711 but most of customers have ADSL connection with 128k upstream, so the result is that they can hear in excellent conditions but can't be heard very well the sound is all choppy. even directly to broadvoice thru Xten sip client. So the idea was to act as proxy and codec converter so that the communication coming out their router is the smaller it can get. I've mentioned G729 or G.723 becuase their routers have it, (Draytek 2600V). Thanks in advance for your suggestions Helder Rogerio___Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to remove + in CIDNumber
got a DID from libretel. Libretel sends CIDnumber as +number ( such as +71898765421) or just number( such as 71898765421) I want to get rid of + before the number, how to setup this in extensions.conf ? ___ 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 forward a call to mobile?
i have an account with BV on my asterisk, how to forward a unanswered incoming call to my mobile phone ( when there is no one to answer the incoming call after 3 rings) ? ___ 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] Spawn extension -----what does this mean ?
for every call, * gives out: Spawn extension (default, 00x, 3) exited non-zero on 'SIP/201.50.117.161-081628e0' Spawn extension -what does this mean ? how to avoid this ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] max call rate (ingress direction) 1.00/30
oh323 show conf : .. max call rate (ingress direction) 1.00/30 --what does1.00/30 mean ? how to increase it ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] invalid frame size for G.729( 2 bytes)
sometime, * gives out tons of : chan_oh323.c:2143 oh323_write: warning : OH323/L1648 invalid frame size for G.729( 2 bytes) chan_oh323.c:2143 oh323_write: warning : OH323/L1648 invalid frame size for G.729( 12 bytes) is there anything wrong ? how to fix it? Mario ___ 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] Light weight and slimmed Asterisk
you can tell * donesn't load these modules in modules.conf On 5/3/05, Kumara Jayaweera [EMAIL PROTECTED] wrote: Greetings to all! Sorry for the numerous postings. but How could I slim my Asterisk PBX. Really I don't need such modules like Ex. chan_modem.so. Becouse, I don't have any special hardware. Please, could I hope your various suggetions in this regards. brief me your idea. ___ 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] how stable is oh323 ?
how stable is oh323 ? is there any production implement ? ( which version 0.65 or 0.7?) could you share the experience ? Mario ___ 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 disconnect a call manually
1 after giving command oh323 show channels, i want to disconnect a call, is there any command to disconnect a call? 2 how asterisk kill a hung/dead call ? for most commercial softswitch, there are a setting for maximum duration for a call. they will hang up it l if its duration reachs the limit. ___ 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] which port is used when asterisk -r
which TCP port is used when asterisk -r ? is there a command to connect to a remote machine ? ( asterisk -r remote-machine-ip ?) ___ 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 set jitter buffer for SIP
there is jitter setting for h323 in oh323.conf where to set min/max jitter buffer for SIP ? i am getting bad voice via *, maybe this jitter buffer setting will help ___ 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 set jitter buffer for SIP
there is jitter setting for h323 in oh323.conf where to set min/max jitter buffer for SIP ? i am getting bad voice via *, maybe this jitter buffer setting will help ___ 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] g729 passthrough?
i am trying to get G723 passthrough get the same error. how to configure passthrough for g723/g729 ? On 4/24/05, Brian Capouch [EMAIL PROTECTED] wrote: jltaylor wrote: ;;; Brian, Add to the [general] section in sip.conf the following: disallow=all allow=g729 allow=ulaw allow=alaw For some reason Asterisk will not pass audio through itself without trying to transcode unless you have this in your config. Don't ask me why it will not work with allow=g729 under the individual peer. This has to go in the [general] section. Still no joy. Added the allow=g729 to general, too, and I still get the same errors. Thanks anyways. B. ___ 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] Any work around for ISPs that block port 5060 and 69
i get the same problem. My option is to change my asterisk box to work completely on a different port, but after set bindport=5061 in sip.conf, asterisk still listens on 5060 after resarting. it seems bindport setting doesnt work. any idear on how to change the sip listening port ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with g729 codec
G729 will not work without a licensecan't G729 work in passthrough mode without license? if yes, how to configure it work in passthrough mode? On Fri, 04 Mar 2005 08:50:11 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Fri, 2005-03-04 at 13:29 +0100, [EMAIL PROTECTED] wrote: Hello, I´m trying the g729 codec for testing pourpose. Whe I try to make a SIP call from a phone using g729 codec to another phone using another codec, when the destination phone answer, the call hangs up. this happend in both ways. In the asterisk console I get. Mar 4 13:11:35 NOTICE[24572]: channel.c:1724 ast_set_write_format: Unable to find a path from gsm to g729 What does it mean? Could this occur cause I am using the g729 without licence? If i buy a licence could solve my problem? G729 will not work without a license. The error message above told you that asterisk couldn't find a valid path to convert from gsm audio to g729 audio data. Seems that should have been very obvious from the error. It is well documented had you even decided to search. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoIP Problems?
www.mutualphone.com On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote: Anyone having problems with LiveVoIP lately? I am seeing failed outgoing calls. Calls that are being routed to wrong numbers. DID's that ring busy. For the pass 2 days I am unable to pass CID. Is anyone else have these problems? Can anyone recommend a Quality VoIP provider? ___ 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] G.729 and mutualphone service
Hi, i am trying to connect to mutualphone.com service. but they only support G.729 /G.723? Must I buy/install G.729 codec at asterisk in order to connect to mutualphone? or I can use the pass-through mode to connect to it without G.729 installed ? if yes, how to configure G.729 passthrough mode? Mario ___ 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
