[Asterisk-Users] Re: Pickup *8 with CallerID
Nik Engel wrote: Hi list ! I implemented *8 to pickup any call on my asterisk system. But after the pickup callerid is missing, so there is no way to see from where the call originated. How can this callerid be passed on. Nik Hi Nik, I'm after the same question as I would like to keep callerID data after pickuping up the call. Maybe using a combination of Pickup and Steal applications would help ? What do you think of this: - You pickup the call, - You then park it to given parking lot with a 2 seconds wait duration and your own extension before call back - You hangup - You're then called back by parking application - You should then see your callerID on your phone's screen. Cheers ___ --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] Timeouts not working
Hi all
I have a problem when im trying to configure a hunt group on zap channel.
here is the part of my extension.conf that not working.
exten = s,1,Answer
exten = s,2,Dial(SIP/[EMAIL PROTECTED],10,Tt)
exten = s,3,GotoIf($[$[${DIALSTATUS} = CHANUNAVAIL] | $[${DIALSTATUS}
= CONGESTION] ]?4)
exten = s,4,Dial(Zap/4/901278**|10)
exten = s,5,Playback('connect-oncall-eng');
exten = s,6,Dial(SIP/${GLOBAL($BridgwaterMB)[EMAIL PROTECTED],20,Tt)
exten = s,7,GotoIf($[$[${DIALSTATUS} = CHANUNAVAIL] | $[${DIALSTATUS}
= CONGESTION] ]?8)
exten = s,8,Dial(Zap/4/9w${GLOBAL($BridgwaterMB)}|20)
exten = s,9,Playback('connect-oncall-eng2')
exten = s,10,Dial(SIP/${GLOBAL($BristolMB)[EMAIL PROTECTED],20,Tt)
exten = s,11,GotoIf($[$[${DIALSTATUS} = CHANUNAVAIL] |
$[${DIALSTATUS} = CONGESTION] ]?12)
exten = s,12,Dial(Zap/4/9w${GLOBAL($BristolMB)}|20)
exten = s,n,Playback(hq)
exten = s,n,Goto(2)
Basically im using Zap/4 as a failover for a SIP trunk when thats not
available
the problem is at s,4 it just dials that number and never times out
any ideas
Cheers
Richard Trenchard
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[asterisk-users] How to handle SIP-Callerid?
Hi, on ISDN there are the numbering plans that indicate if it's an national or an internation number. Is there something similar on SIP? How should i set a callerid to an internation number? complete e164, with, without an intl prefix (ie +, 011, 00 etc)...? How to a national number? Regards, Andreas. _ Discover fun and games at @ http://xtramsn.co.nz/kids ___ --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] Queue Announcements for Operators
Hi All I would like to be able to have an announcement played to an operator advising them of the queue the call came from before the call is pasted over to them, so they know how to greet the customer. Does anyone have any ideas or can point me to some resource which details this? Many Thanks in Advance. SP ___ --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] Info: Nokia E65 working with Asterisk
Hi, Just for information on compatibility: Earlier this week I got a Nokia E65 which supports WiFi and SIP. I got the WiFi side configured to work with an access point after several attempts. This eventually had to be done using all manual settings, as using it's config wizard gave WEP Key errors despite many attempts and careful verification. It's WiFi system is not particularly sensitive, it does not detect an AP at the far side of the building that is quite useable from the this location by a typical notebook PC. This may be an advantage, as it's not going to connect to anything with a weak signal that could drop out with movement. I set up a standard SIP extension for it in asterisk via freepbx. Some of the phone's SIP settings are less than obvious, at least to me (i.e. where to put the Asterisk box's IP), but I found an excellent guide here: http://newlc.com/Using-SIP-with-Nokia-Series60-and.html After adjusting the settings in line with that, it works perfectly. I've also set to be permanently registered so it's available for incoming calls while in range. I have left the default for outgoing calls to be the mobile network. To make a call via the Asterisk PBX, you need to enter the number then press the 'options' key, select 'Call' go to 'Internet Call'. Voice quality is excellent and I've not had any problems with it so far. Robert Jenkins. ___ --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] pritimer parameter in zapata.conf
Hi all, Please discribe me more about pritimer parameter in zapata.conf http://lists.digium.com/pipermail/asterisk-commits/2006-July/005824.html I found above url and have some idea. My PRI E1 timer is t203, what is the best vale that i have to use for as counter. default is 1ms, If i changed it to some big amount, like 6 what will happen T203: Layer 2 max time without frames being exchanged (default 1 ms) -- Thanks Regards, Vidura B. Senadeera. ___ --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] Empty Wildcard TDM400P as a MeetMe timer.
I've just moved into 3.3v PCI servers and found that my clone X100P cards were lying about the 3.3v supported notch. Can I use a Wildcard TDM400P without any modules as a timer for MeetMe in a 64 bit 3.3v server? Will I still need to plug the hard disk power cable into it? Is there a better cheaper 3.3v MeetMe timer? (Boss doesn't trust the kernel timer.) -HJC ___ --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: build rpm fails
Axel Thimm wrote: As fast as they read asterisk-announce ;) I doubt that you are that fast ;) but I thank you for answer. -- Tomislav Parcina [EMAIL PROTECTED] ___ --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 queue and agents
What version of Asterisk is this the r number on the 1.4 branch? I'll try and reproduce the condition here. Also - if you could post into that bug on Mantis a full DEBUG/VERBOSE log and what it looks like when you do show queues when one of these agents is on the phone, that'd be real helpful. Thanks. On 3/7/07, Hall, Eric M. [EMAIL PROTECTED] wrote: BJ Here is the sip.conf file. Hints work great. The only problem is the queue is sending calls to an agent that's on the phone. [general] rtcachefriends=yes videosupport=yes port=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) context=sip ; Send unknown SIP callers to this context allow=g729 allow=h263 ; H.263 is our video codec allow=h263p ; H.263p is the enhanced video codec ;allow=g711 ;allow=all ;allow=ulaw ;allow=gsm nat=1 host=dynamic type=peer qualify=yes notifyringing=yes Subscribecontext=sip call-limit=300 notifyhold = yes limitonpeer = yes notifyringing = yes; Notify subscriptions on RINGING state (default: no) notifyhold = yes [56405] ;Eric Test type=friend ; friend means this device takes and makes calls username=1 ; Username on device callerid=Eric Test Phone 56405 secret=56405; Password for device host=dynamic ; This host is not on the same IP addr every time context=sip ; Inbound calls from this host go here [EMAIL PROTECTED]; Activate the message waiting light if this canreinvite=no; Leave this alone for now; see archives for details nat=1 qualify=yes Subscribecontext=sip notifyringing=yes call-limit=300 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, March 07, 2007 10:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents I don't think this is a bug. From UPGRADE.txt: * Queues depend on the channel driver reporting the proper state for each member of the queue. To get proper signalling on queue members that use the SIP channel driver, you need to enable a call limit (could be set to a high value so it is not put into action) and also make sure that both inbound and outbound calls are accounted for. Example: [general] limitonpeer = yes [peername] type=friend call-limit=10 Please test with that and report your findings, and if it's still not working find us on IRC as we'd like to take a further look and see what might be wrong. BJ On 3/7/07, Hall, Eric M. [EMAIL PROTECTED] wrote: Looks like it's a bug http://bugs.digium.com/view.php?id=9172nbn=3 I have update to Asterisk SVN-branch-1.4-r58243 and will test it tomorrow and report back to the list. Eric Hall -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavio Ruiz (Ta^3) Sent: Wednesday, March 07, 2007 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents Have a question for the group If I have an agent is on the phone outside of the queue should that person still get queue calls ? Doing a show agents online I see Available however show hints I see inuse. There is a ringinuse feature for SIP devices on 1.4.X which is what you are looking for. -- Octavio Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Cel.: (+55 55) 5514-087790 ___ --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 -- 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 ___ --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 -- 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
[asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk
Matt wrote: Thanks I was just about to say this. You CAN'T send caller-id-name. To be able to set name you need to set it with Telcordia or whomever manages numbers in your country. Optima provider in Croatia allows users to set up CallerID name on outgoing PRI calls. -- Tomislav Parcina [EMAIL PROTECTED] ___ --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: queue information into db
Why don't we start a cvs? On 3/8/07, David Boyd [EMAIL PROTECTED] wrote: Thank you very much, as we make changes or modifications we will keep you posted. Dave On Thu, 2007-03-08 at 08:43 +0100, nik600 wrote: https://sourceforge.net/projects/ccmanager/ please note that it is a beta version, i'd like to improve it but i'm busy with work and university. take a look and let me know. nik On 3/6/07, nik600 [EMAIL PROTECTED] wrote: i've submittet the project to SF. I have to wait 2 business days for their validation. The project is in a beta release and will be released on GPL. Bye On 3/2/07, Tomislav Parcina [EMAIL PROTECTED] wrote: nik600 wrote: i'm sorry but due to some problem the software will be released not first than Wednesday 7/02/2007. i'll post a message . This should be Wednesday 7/3/2007. right? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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 queue and agents
Asterisk SVN-branch-1.4-r58243 Voipgw*CLI show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online , 13 offline] voipgw*CLI show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online , 13 offline] If you tell me how to do a full DEBUG/VERBOSE I will be happy to send you one. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Thursday, March 08, 2007 7:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents What version of Asterisk is this the r number on the 1.4 branch? I'll try and reproduce the condition here. Also - if you could post into that bug on Mantis a full DEBUG/VERBOSE log and what it looks like when you do show queues when one of these agents is on the phone, that'd be real helpful. Thanks. On 3/7/07, Hall, Eric M. [EMAIL PROTECTED] wrote: BJ Here is the sip.conf file. Hints work great. The only problem is the queue is sending calls to an agent that's on the phone. [general] rtcachefriends=yes videosupport=yes port=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) context=sip ; Send unknown SIP callers to this context allow=g729 allow=h263 ; H.263 is our video codec allow=h263p ; H.263p is the enhanced video codec ;allow=g711 ;allow=all ;allow=ulaw ;allow=gsm nat=1 host=dynamic type=peer qualify=yes notifyringing=yes Subscribecontext=sip call-limit=300 notifyhold = yes limitonpeer = yes notifyringing = yes; Notify subscriptions on RINGING state (default: no) notifyhold = yes [56405] ;Eric Test type=friend ; friend means this device takes and makes calls username=1 ; Username on device callerid=Eric Test Phone 56405 secret=56405; Password for device host=dynamic ; This host is not on the same IP addr every time context=sip ; Inbound calls from this host go here [EMAIL PROTECTED]; Activate the message waiting light if this canreinvite=no; Leave this alone for now; see archives for details nat=1 qualify=yes Subscribecontext=sip notifyringing=yes call-limit=300 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, March 07, 2007 10:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents I don't think this is a bug. From UPGRADE.txt: * Queues depend on the channel driver reporting the proper state for each member of the queue. To get proper
Re: [asterisk-users] Info: Nokia E65 working with Asterisk
I have left the default for outgoing calls to be the mobile network. To make a call via the Asterisk PBX, you need to enter the number then press the 'options' key, select 'Call' go to 'Internet Call'. Is this 'Call' go to 'Internet Call' usable when you select a callee using the phone's directory ? ___ --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 queue and agents
Sorry Forgot to tell you I was on exten 56405 called to my cell. I then called into the Queue with another cell and this is the output. Also forgot to include the show queue voipgw*CLI show queue dayton has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: agent/56432 (Unavailable) has taken no calls yet agent/56422 (Unavailable) has taken no calls yet agent/56426 (Unavailable) has taken no calls yet agent/56424 (Unavailable) has taken no calls yet agent/56429 (Unavailable) has taken no calls yet agent/56427 (Unavailable) has taken no calls yet agent/56425 (Unavailable) has taken no calls yet agent/56411 (Unavailable) has taken no calls yet agent/56428 (Unavailable) has taken no calls yet No Callers masion has 1 calls (max unlimited) in 'fewestcalls' strategy (0s holdtime), W:0, C:0, A:2, SL:0.0% within 0s Members: agent/564321 (Unavailable) has taken no calls yet agent/564221 (Unavailable) has taken no calls yet agent/56405 (paused) (Not in use) has taken no calls yet agent/56423 (Unavailable) has taken no calls yet agent/56421 (paused) (Not in use) has taken no calls yet agent/56420 (Unavailable) has taken no calls yet agent/56416 (paused) (Not in use) has taken no calls yet Callers: 1. SIP/208.70.216.73-09780030 (wait: 0:12, prio: 0) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Thursday, March 08, 2007 7:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk queue and agents Asterisk SVN-branch-1.4-r58243 Voipgw*CLI show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online , 13 offline] voipgw*CLI show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online , 13 offline] If you tell me how to do a full DEBUG/VERBOSE I will be happy to send you one. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Thursday, March 08, 2007 7:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents What version of Asterisk is this the r number on the 1.4 branch? I'll try and reproduce the condition here. Also - if you could post into that bug on Mantis a full DEBUG/VERBOSE log and what it looks like when you do show queues when one of these agents is on the phone, that'd be real helpful. Thanks. On 3/7/07, Hall, Eric M. [EMAIL PROTECTED] wrote: BJ Here is the sip.conf file. Hints work great. The only
[asterisk-users] Re: visdn, misdn and the hell
Massimo Nuvoli wrote: I think the ISDN part of asterisk is very important, in Italy there is a lot of equipments that are ISDN and not ANALOGIC or PRI, and with no ISDN stable support it is impossibile to port asterisk on the real world. In Croatia also. Small companies are just to small for PRI and they all use ISDN BRI lines. Wath i see now is that a lot of integrators are doing this: using external box to avoid at 100% the isdn problem in asterisk. Very bad, we go to use proprietary designed hardware and software, external components, more complexity, more point of failure. Definitely agree with you. -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] Info: Nokia E65 working with Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 8 Mar 2007, at 13:34, Olivier wrote: I have left the default for outgoing calls to be the mobile network. To make a call via the Asterisk PBX, you need to enter the number then press the 'options' key, select 'Call' go to 'Internet Call'. Is this 'Call' go to 'Internet Call' usable when you select a callee using the phone's directory ? Yes it is. However, this also depends on how you set up your dial plan and how you store phone numbers in your directory. I have set up my Asterisk dial plan to understand and work with the universal phone number notation of +country codearea codenumber, which is understood by the mobile network as well. I store all my phone numbers that way, be they local, long distance or international long distance from where I am. This means I can select any phone number from my phone book and dial out via the mobile network or my Asterisk server, it just works. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFF8AnFRAx5nvEhZLIRAqPbAKCH2IxZAvTTtt4D8WjbzU5WVz6FGACfTVD6 bAaLd67dNaiatajZ3nSdP4A= =V36x -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] Asterisk queue and agents
Ok. One more thing - how are you logging the agent in? With AgentLogin or AgentCallBackLogin? Additionally, how did you get on that call 56405 to your cell? Was it directly to the SIP device or via the agent channel that the represents that SIP device? BJ On 3/8/07, Hall, Eric M. [EMAIL PROTECTED] wrote: Sorry Forgot to tell you I was on exten 56405 called to my cell. I then called into the Queue with another cell and this is the output. Also forgot to include the show queue voipgw*CLI show queue dayton has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: agent/56432 (Unavailable) has taken no calls yet agent/56422 (Unavailable) has taken no calls yet agent/56426 (Unavailable) has taken no calls yet agent/56424 (Unavailable) has taken no calls yet agent/56429 (Unavailable) has taken no calls yet agent/56427 (Unavailable) has taken no calls yet agent/56425 (Unavailable) has taken no calls yet agent/56411 (Unavailable) has taken no calls yet agent/56428 (Unavailable) has taken no calls yet No Callers masion has 1 calls (max unlimited) in 'fewestcalls' strategy (0s holdtime), W:0, C:0, A:2, SL:0.0% within 0s Members: agent/564321 (Unavailable) has taken no calls yet agent/564221 (Unavailable) has taken no calls yet agent/56405 (paused) (Not in use) has taken no calls yet agent/56423 (Unavailable) has taken no calls yet agent/56421 (paused) (Not in use) has taken no calls yet agent/56420 (Unavailable) has taken no calls yet agent/56416 (paused) (Not in use) has taken no calls yet Callers: 1. SIP/208.70.216.73-09780030 (wait: 0:12, prio: 0) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Thursday, March 08, 2007 7:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk queue and agents Asterisk SVN-branch-1.4-r58243 Voipgw*CLI show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online , 13 offline] voipgw*CLI show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online , 13 offline] If you tell me how to do a full DEBUG/VERBOSE I will be happy to send you one. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Thursday, March 08, 2007 7:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents What version of Asterisk is this the r number on the 1.4 branch? I'll try and reproduce the condition here. Also -
RE: [asterisk-users] Asterisk queue and agents
I use AgentCallBackLogin I called that exten from my cell. However I have tested it calling into the Queue with the same outcome. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Thursday, March 08, 2007 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents Ok. One more thing - how are you logging the agent in? With AgentLogin or AgentCallBackLogin? Additionally, how did you get on that call 56405 to your cell? Was it directly to the SIP device or via the agent channel that the represents that SIP device? BJ On 3/8/07, Hall, Eric M. [EMAIL PROTECTED] wrote: Sorry Forgot to tell you I was on exten 56405 called to my cell. I then called into the Queue with another cell and this is the output. Also forgot to include the show queue voipgw*CLI show queue dayton has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: agent/56432 (Unavailable) has taken no calls yet agent/56422 (Unavailable) has taken no calls yet agent/56426 (Unavailable) has taken no calls yet agent/56424 (Unavailable) has taken no calls yet agent/56429 (Unavailable) has taken no calls yet agent/56427 (Unavailable) has taken no calls yet agent/56425 (Unavailable) has taken no calls yet agent/56411 (Unavailable) has taken no calls yet agent/56428 (Unavailable) has taken no calls yet No Callers masion has 1 calls (max unlimited) in 'fewestcalls' strategy (0s holdtime), W:0, C:0, A:2, SL:0.0% within 0s Members: agent/564321 (Unavailable) has taken no calls yet agent/564221 (Unavailable) has taken no calls yet agent/56405 (paused) (Not in use) has taken no calls yet agent/56423 (Unavailable) has taken no calls yet agent/56421 (paused) (Not in use) has taken no calls yet agent/56420 (Unavailable) has taken no calls yet agent/56416 (paused) (Not in use) has taken no calls yet Callers: 1. SIP/208.70.216.73-09780030 (wait: 0:12, prio: 0) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Thursday, March 08, 2007 7:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk queue and agents Asterisk SVN-branch-1.4-r58243 Voipgw*CLI show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online , 13 offline] voipgw*CLI show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online
Re: [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.
Henry Cobb wrote: I've just moved into 3.3v PCI servers and found that my clone X100P cards were lying about the 3.3v supported notch. Can I use a Wildcard TDM400P without any modules as a timer for MeetMe in a 64 bit 3.3v server? Will I still need to plug the hard disk power cable into it? Is there a better cheaper 3.3v MeetMe timer? (Boss doesn't trust the kernel timer.) -HJC ___ The empty card will work but there is some trickery that must be done. I asked this same question about a year ago, some people said it will not work and one person said it would and gave me the code change. It worked. I forget what the change was but you should be able to search the archives for my name plus empty tdm400p or something like that. Thanks, Steve Totaro ___ --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 Auto-dial out
Perfect! Thanks a lot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Wednesday, March 07, 2007 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk Auto-dial out I am using the * auto-dial out feature but don't want to have to specify a channel (Zap/G2/) to connect to the extension. Current file I use: Channel: Zap/G2/12127778866 # I have to specify a specific channel MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [line1out] # Context: line1out Extension: 7632 Priority: 1 Is there a way that I can just put in the number and have the system decide the channel to use for calling it? What I would like to do: Channel: #=== This number could be # 7645 in which case go via SIP/7645 # 68001 which should go to CiscoSIP/68001 # 12127778866 which would go via Zap/G2/12127778866 MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [line1out] # Context: line1out Extension: 7632 Priority: 1 Based on dialing plan the system should be able to route the call to whatever channel supports dialing that number. You probably want to use the Local channel. Definitely hit the wiki and check it out: http://www.voip-info.org/wiki/view/Asterisk+local+channels The idea behind the local channel is that you can, in effect, drop a call right into a specific part of the dialplan. From there, your dialplan can handle the logic of figuring out which technology and channel to use. -MC ___ --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] Hinting and Realtime
hello all, My problem if i have my extensions and sipusers in a realtime database it is not possible to use BLF or hinting. i see only idle or unavailable status but if the phone is ringing or in use i can't see it. Is there a fix or any workaround? Version is Release 1.4.1 regards rene ___ --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] Empty Wildcard TDM400P as a MeetMe timer.
Steve Totaro wrote: Henry Cobb wrote: I've just moved into 3.3v PCI servers and found that my clone X100P cards were lying about the 3.3v supported notch. Can I use a Wildcard TDM400P without any modules as a timer for MeetMe in a 64 bit 3.3v server? Will I still need to plug the hard disk power cable into it? Is there a better cheaper 3.3v MeetMe timer? (Boss doesn't trust the kernel timer.) -HJC ___ The empty card will work but there is some trickery that must be done. I asked this same question about a year ago, some people said it will not work and one person said it would and gave me the code change. It worked. I forget what the change was but you should be able to search the archives for my name plus empty tdm400p or something like that. Thanks, Steve Totaro It was a little harder to find than I thought so I will give you a link to the thread. http://lists.digium.com/pipermail/asterisk-users/2006-May/151222.html Thanks, Steve Totaro www.asteriskhelpdesk.com KB3OPB ___ --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 + Panasonic pbx
Sir, Please help me how to connect asterisk pbx having FXS port with panasonic pbx. Rajeev. ___ --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 Linksys SPA Daylight Saving Time Rule for US/Canada
To work with the latest change to the US/Canadian DST, I made a new Daylight Saving Time Rule for my Linksys SPA-9XX phones. start=3/7/7/02:00:00;end=11/1/7/02:00:00;save=1 As I could see no way to tell the phones to begin DST on the second Sunday in March, I assumed that the second Sunday would always be at least on or after the 7th of the month. Let me know if you see any obvious flaws to my logic, or the rule itself. Thanks. Sincerely, Trevor Hammonds ___ --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] Queue Announcements for Operators
scott wrote: I would like to be able to have an announcement played to an operator advising them of the queue the call came from before the call is pasted over to them, so they know how to greet the customer. Does anyone have any ideas or can point me to some resource which details this? That sounds like the announce option in the sections in queues.conf would solve you problem. ---cut--- ; An announcement may be specified which is played for the member as ; soon as they answer a call, typically to indicate to them which queue ; this call should be answered as, so that agents or members who are ; listening to more than one queue can differentiated how they should ; engage the customer ; ;announce = queue-markq ---cut--- Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Hinting and Realtime
René Enskat wrote: My problem if i have my extensions and sipusers in a realtime database it is not possible to use BLF or hinting. i see only idle or unavailable status but if the phone is ringing or in use i can't see it. Is there a fix or any workaround? Version is Release 1.4.1 Hints do not work with Realtime. So the solution would probably be not to have the extensions in the database. Someone please prove me wrong. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Fwd: Back to back E1 - asterisk = toshiba pbx -Call droping
Before studying your configs, what have you tried so far? Did you change this? Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4 to span=2,0,0,ccs,hdb3,crc4. Here is the documentation on voip-info for why it may be the cause of your issues http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax span definition format: span=(spannum),(timing),(LBO),(framing),(coding) spannum= Number of the span. timing= How to synchronize the timing devices. 0: to not use this span as sync source 1: to use as primary sync source 2: to set as secondary and so forth Use '1' if you want to use the circuit as your primary sync source. If '0' is used asterisk will try to provide timing to the span (say, if you were connecting to a legacy PBX). If Asterisk is connected directly to the telco you will want to use '1' to accept timing from them. If youhave multiple spans, set them as 2, 3, 4, etc. Problems with timing manifest themselves different ways - with static, pops, and channels or calls regularly dropping. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vidura Senadeera Sent: Thursday, March 08, 2007 1:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Fwd: Back to back E1 - asterisk = toshiba pbx -Call droping -- Forwarded message -- From: Vidura Senadeera [EMAIL PROTECTED] Date: Mar 8, 2007 11:27 AM Subject: Re: Back to back E1 - asterisk = toshiba pbx - Call droping To: asterisk-users@lists.digium.com Hi steve and All, I'm attaching cat proc/interrupts out put, lspci -bv out put, zapa.conf, zaptel.conf for your information Thanks so much for the feedback and I do accordingly. Hope to get rid off this isue any how. To day also reported 10 call drops within 2 hours of period. fook forward to have your support on this regard. Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk http://www.debug.lk/ Message: 16 Date: Wed, 7 Mar 2007 05:05:36 -0500 From: Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Subject: RE: [asterisk-users] Back to back E1 - asterisk = toshiba pbx - Calldroping issue To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] m mailto:[EMAIL PROTECTED] pdesk.com Content-Type: text/plain; charset=us-ascii As these problems are very time sensitive and frustrating, I suggest you document each change you make and do them one at a time so you can actually know what the problem was and not introduce new problems in the process. Find someone who is on the phone quite a bit and will give you an honest evaluation of the call dropping situation (unless you yourself are experiencing this issue too). Some people are so quick to say, It is still happening without starting the evaluation from a clean slate after each change. You may want to check your Asterisk log for more insight. /var/log/asterisk/full. Also you can turn on debugging on one span at a time and see if you can find something there Do you have a resetinterval set in zapata.conf? If you can isolate the dropped calls to the reset interval (watch the console, it will scroll with each channel being reset) then set resetinterval=never. If there is no entry for resetinterval, add it and set it to never since it is defaulted to on. Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4 to span=2,0,0,ccs,hdb3,crc4. This in combination with your first span should accept timing from the Telco and then supply it to your Toshiba, I would actually try this first. Another thought, It seems you have quite a lot of hardware in that box. I am not sure how much is too much, but that would probably just rear it's ugly head as poor audio. Thanks, Steve Totaro http://www.asteriskhelpdesk.com http://www.asteriskhelpdesk.com/ _ From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] On Behalf Of Vidura Senadeera Sent: Wednesday, March 07, 2007 2:15 AM To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: [asterisk-users] Back to back E1 - asterisk = toshiba pbx - Calldroping issue Hi
AW: [asterisk-users] Hinting and Realtime
But with 1.2.x it is working No big voip-carrier will have 1000 accounts in a file. So there must be an implementation for that again. Regards rene -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Philipp Kempgen Gesendet: Donnerstag, 8. März 2007 15:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Hinting and Realtime René Enskat wrote: My problem if i have my extensions and sipusers in a realtime database it is not possible to use BLF or hinting. i see only idle or unavailable status but if the phone is ringing or in use i can't see it. Is there a fix or any workaround? Version is Release 1.4.1 Hints do not work with Realtime. So the solution would probably be not to have the extensions in the database. Someone please prove me wrong. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Queue Announcements for Operators
www.voip-info.org ; Announcement to be played to an agent answering a call. ; This is intended so that agents that are members of more than one queue can ; determine how to greet callers. ;announce = queue-support Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of scott Sent: Thursday, March 08, 2007 5:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue Announcements for Operators Hi All I would like to be able to have an announcement played to an operator advising them of the queue the call came from before the call is pasted over to them, so they know how to greet the customer. Does anyone have any ideas or can point me to some resource which details this? Many Thanks in Advance. SP ___ --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] AMI Originate and release channels
Hi, I put /n option, but still not working msg += Channel: Local/[EMAIL PROTECTED]/n\r\n But the Local Channel doesn't hangs up... Any idea? tks Paulo 2007/2/8, Steve Murphy [EMAIL PROTECTED]: On Thu, 2007-02-08 at 10:32 -0200, Paulo Vicentini wrote: Hi I set up call back functionally thru AMI (local channel). The two calls are bridged and the call is established. But when I hang up the local channel (the first extension that rang), the other leg of the call *is not released* Are you using the /n option with the local channel spec? ie, Local/.../n ? murf ___ --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: AW: [asterisk-users] Hinting and Realtime
René Enskat wrote: But with 1.2.x it is working No big voip-carrier will have 1000 accounts in a file. So there must be an implementation for that again. Regards rene -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Philipp Kempgen Gesendet: Donnerstag, 8. März 2007 15:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Hinting and Realtime René Enskat wrote: My problem if i have my extensions and sipusers in a realtime database it is not possible to use BLF or hinting. i see only idle or unavailable status but if the phone is ringing or in use i can't see it. Is there a fix or any workaround? Version is Release 1.4.1 Hints do not work with Realtime. So the solution would probably be not to have the extensions in the database. Someone please prove me wrong. Regards, Philipp The fact that realtime extensions and hints worked with 1.2 is new to me. Anyone else? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] cmd pickup Problem
Hi there, i have a Problem with the Pickup command. Versions: asterisk 1.4.1 on gentoo my extensions.conf [only the interesting part]: [incoming_1] exten = 123,1,Ringing exten = 123,2,Dial(SIP/,20,r) exten = 123,3,wait(90) exten = 123,4,hangup [incoming_2] exten = 456,1,pickup([EMAIL PROTECTED]) both are sip-accounts and have pickupgroup=1 in the sip.conf so my idea is, when anybody calls at 123 my mobile is ringing and i call back on 456 and will be connected to the caller the callout and all other are runnig, but at the pickup there is always: pickup_exec: No target channel found for [EMAIL PROTECTED] I've already tried to insert a answer before the pickup and do a pickup without the context but nothing runs... any ideas why my pickup don't it? thanx Juergen ___ --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 + Panasonic pbx
Look at options on www.voip-info.org http://www.voip-info.org/wiki/index.php?comment_page=1page_id=566maxComments=1comments_maxComments=1comments_sort_mode=commentDate_asccomments_style=flat Thanks, Steve Totaro Sanspareils Greenlans wrote: Sir, Please help me how to connect asterisk pbx having FXS port with panasonic pbx. Rajeev. ___ --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] Asterisk distributed deployment
Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like: Asterisk servers on all locations(central and remote offices) or Asterisk on Central office plus FXO Gateways on remote offices, all of this connected through a central asterisk cluster With the first option i have TDM cards seller that offer me DIGIUM (expensive) or OPENVOX (less expensive), but because i not have experience with OPENVOX telephony hardware I cant consider that. So, if Any can give me some good reasons for use OPENVOX against DIGIUM cards i would have solve this question because may build IAX trunks on each office. With the 2nd option I have sellers that offer me gateways: Quintum Tenor AFT400 Planet VIP-480 FO But, again, I don't have experience with asterisk and FXO gateways to think that it is the best solution amen that is the less expensive solution. Another solution that i consider is mount asterisk on central office and IP PBX DIGISTAR preconfigured on remote offices. On the Users Side I was considering the use of Ata's or FXS Gateways, with Ata's I get offers of Audiocodes MP202, GRANDSTREAM HT 386 or Linksys SPA-2002. And with FXS Gateways sellers offers me Quintum Tenor AXG2400, Quintum Tenor AFG800 Thanks for any word that can help me to get this VoIP deployment working and sorry for my english. Cheers G. ___ --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] Call recording and archiving
Voip Asterisk wrote: Does anyone have a good suggestion for a automated solution to record calls on certain interfaces and easily archiving them in a way which is easily matched against CDRs? Also can someone suggest the appropriate protocol to archive the recording when the conversations are transpiring in ulaw. Basically a nice cost effective trade off between CPU and disk space for medium call load. Miles, I believe that you should be able to name a recorded file so that it contains a unique value that ties it back to its CDR. Read 'doc/README.cdr' for information on customizing your CDRs, and 'show application monitor' at the Asterisk CLI for documentation on changing the recording's filename. Hopefully, one of the more knowledgeable list members will correct me if I'm wrong and fill in the gaps I've left. The choice of a codec for the recordings is debatable, but here is what we're doing. All of our calls are u-law and are recorded locally on the Asterisk server as two PCM leg files. We pass the Monitor application the 'm' flag, which tells it to mix the leg files at the end of the call. To perform the mixing, Asterisk calls soxmix by default. We have replaced the soxmix binary with a script that moves the leg files across an NFS mount to our digital recording server (MONITOR_EXEC could be used for this, but we found it to be unreliable). A process on the digital recording server sweeps for new recordings, mixes them as GSM WAVs, and indexes them for retrieval. As I said, the codec is debatable. We chose GSM, because it has a decent compression ratio, handles voice well, and plays on most media players without the need to install additional codecs on the machine. The NFS mount and the separate server for mixing, indexing, and archiving recordings are not necessary if you have a relatively low call load but I highly recommend it on a busy machine. Transcoding is a CPU intensive task, so its a great candidate for offloading. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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: Asterisk Realtime
I enabled some more detailed debugging and logging as per someone else a few posts ago and I saw that the permissions on MySQL were set incorrectly. I granted all, but what are the least permissions this user should need? How do I register to other servers? It seems to be ignoring the register statements in my iax.conf. --Mike All that looks fine. What do you get when you do realtime mysql status? The next areas to look at would be your DB configs, and debug status when you actually try to use one of the entries in your DB. . . I only use it for iaxpeers/users and extensions, so I can't comment much on its use with SIP or voicemail. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 distributed deployment
[EMAIL PROTECTED] wrote: Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like: Asterisk servers on all locations(central and remote offices) or Asterisk on Central office plus FXO Gateways on remote offices, all of this connected through a central asterisk cluster With the first option i have TDM cards seller that offer me DIGIUM (expensive) or OPENVOX (less expensive), but because i not have experience with OPENVOX telephony hardware I cant consider that. So, if Any can give me some good reasons for use OPENVOX against DIGIUM cards i would have solve this question because may build IAX trunks on each office. With the 2nd option I have sellers that offer me gateways: Quintum Tenor AFT400 Planet VIP-480 FO But, again, I don't have experience with asterisk and FXO gateways to think that it is the best solution amen that is the less expensive solution. Another solution that i consider is mount asterisk on central office and IP PBX DIGISTAR preconfigured on remote offices. On the Users Side I was considering the use of Ata's or FXS Gateways, with Ata's I get offers of Audiocodes MP202, GRANDSTREAM HT 386 or Linksys SPA-2002. And with FXS Gateways sellers offers me Quintum Tenor AXG2400, Quintum Tenor AFG800 Thanks for any word that can help me to get this VoIP deployment working and sorry for my english. Cheers G. So what exactly are you asking? If you want to do it right, put asterisk servers at all of your locations and connect them to the PSTN then use LCR in your dialplan to route calls within the organization and also route calls out the best office as far as TDM rates. Deploy SIP phones that point to their local Asterisk server. Then if your network goes down at any point, each site is fully independent. If you want to go cheap, you have a VPN, setup the main office with an asterisk box with PSTN connectivity and then deploy SIP phones at the remote offices that point over the VPN to your main office. Thanks, Steve ___ --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] Accessing Voicemail by dialing own number
Is is possible to check voicemail by dialing one's own number? When the outgoing voicemail message begins, I'd like to be able to press some key and have it prompt to enter the password for that box. Is this possible, and what option do I need to enable to make this function? ___ --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] Accessing Voicemail by dialing own number
As soon as the vm answers, press *. That's the default I believe to enter VM on that line D. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Carey Sent: Thursday, March 08, 2007 12:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Accessing Voicemail by dialing own number Is is possible to check voicemail by dialing one's own number? When the outgoing voicemail message begins, I'd like to be able to press some key and have it prompt to enter the password for that box. Is this possible, and what option do I need to enable to make this function? ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.8/714 - Release Date: 3/8/2007 10:58 AM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.8/714 - Release Date: 3/8/2007 10:58 AM ___ --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: Asterisk Realtime
Mike Hammett wrote: I enabled some more detailed debugging and logging as per someone else a few posts ago and I saw that the permissions on MySQL were set incorrectly. I granted all, but what are the least permissions this user should need? select, insert, update, delete? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Accessing Voicemail by dialing own number
Yes, you can setup * key to do that, its a standard feature see the docs of the voicemail application for details on how to do it. ___ --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] Accessing Voicemail by dialing own number
Chris Carey wrote:
Is is possible to check voicemail by dialing one's own number?
You could check if ${EXTEN} matches ${CALLERID(num)} and
if so send them to VoicemailMain()
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
___
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[asterisk-users] Queue announcing hold sequence instead of hold time
Hi,
We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian
Sarge) and the behaviour of our Call Centre queues has changed slightly.
Before the upgrade, when a caller was waiting in the queue, the
estimated hold time was announced as expected (estimated hold time is
less than 2 minutes ...).
Now the caller gets an announcement of their sequence in the queue
(Your call is now first in line ...).
I believe that the only changes I have made to queues.conf and
agents.conf is the addition of the context= statement and editing the
list of agents.
Has anyone else seen this? What am I missing?
regards,
Drew
QUEUES.CONF
[general]
persistentmembers = yes
[FxQueue]
music=default
strategy=rrmemory
context = opt-out_fxq
timeout=15
retry=3
wrapuptime=0
maxlen=0
announce-frequency=60
announce-holdtime = yes
reportholdtime=yes
memberdelay=1
servicelevel=120 ; seconds
member = Agent/1102
member = Agent/1103
member = Agent/1104
member = Agent/1105
AGENTS.CONF
[general]
persistentagents=yes
[agents]
ackcall=no
wrapuptime=0
musiconhold = default
agent = 1102,1234,Carly
agent = 1103,1234,Sean
agent = 1104,1234,Ed
agent = 1105,1234,Neil
EXTENSIONS.CONF (extract)
[call-centre]
exten = s,1,Noop(Entering Call Centre)
exten = s,n,Answer()
exten = s,n,Wait(1)
exten = s,n,Playback(welcome-fxqueue)
exten = s,n,Goto(5210,1)
; FxQueue
exten = 5210,1,Noop()
exten = 5210,n,Ringing()
exten = 5210,n,Wait(2)
exten = 5210,n,Queue(FxQueue|tH)
exten = 5210,n,Hangup()
exten = _6XXX,1,macro(ccexten,${EXT_${EXTEN}})
[macro-ccexten]
exten = s,1,Set(EXT=${ARG1})
exten = s,2,GotoIf([$:{EXT}]?4:3)
exten = s,3,Goto(i,1)
exten = s,4,Dial(${EXT},10,tT)
exten = i,1,Playback(pbx-invalid)
--
Drew Gibson
Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com
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[asterisk-users] Packet2Packet Bridging Questions
I'm just starting to upgrade some boxes from 1.2.x to 1.4.1 as well as trying to get some of the RTP traffic offloaded from the network. I think I'm misunderstanding what the console messages mean when it says Packet2Packet Bridding SIP/blah to SIP/blah. I though that meant that it had successfully (re)INVITED and the media was no longer going through my Asterisk box, but ethereal says different. I'm not having much luck finding any information on this on the wiki or google. Can someone point me in the right direction? Thanks, Daryl ___ --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: Asterisk distributed deployment
Steve, Im not asking but looking for a suggest about multiple solutions to the same problem, Im looking for experinces with hibrid deployments that save me money, for example sellers offers me TDM04B DIGIUM CARDS about u$s 500 against u$s 150 for OPENVOX CARDS. Cheers ___ --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] Packet2Packet Bridging Questions
Daryl Jurbala wrote: I'm just starting to upgrade some boxes from 1.2.x to 1.4.1 as well as trying to get some of the RTP traffic offloaded from the network. I think I'm misunderstanding what the console messages mean when it says Packet2Packet Bridding SIP/blah to SIP/blah. I though that meant that it had successfully (re)INVITED and the media was no longer going through my Asterisk box, but ethereal says different. I'm not having much luck finding any information on this on the wiki or google. Can someone point me in the right direction? Thanks, Daryl Packet2Packet Bridging = Audio is not going through the Asterisk core, it comes into the RTP stack and goes directly out. This decreases the amount of memory allocation that happens, and things require less processing. Native Bridging = Audio was reinvited between the two endpoints so it (should) go direct. Joshua Colp Software Developer Digium, Inc. ___ --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: Asterisk distributed deployment
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, March 08, 2007 12:36 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisk distributed deployment Steve, Im not asking but looking for a suggest about multiple solutions to the same problem, Im looking for experinces with hibrid deployments that save me money, for example sellers offers me TDM04B DIGIUM CARDS about u$s 500 against u$s 150 for OPENVOX CARDS. Cheers I think you will pay in the long run if you are going to skimp on U$D350. Couldn't one dropped call from a prospective big time customer be worth more than U$D350? Would spending a week of time trying to get things working correctly be worth U$D350? I would suggest doing it right and not skimping. Your phone system should be transparent and is a direct reflection on your business. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --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] Queue announcing hold sequence instead of hold time
I also have this problem. Unsure how to fix it though.
Rob
Drew Gibson wrote:
Hi,
We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on
Debian Sarge) and the behaviour of our Call Centre queues has changed
slightly.
Before the upgrade, when a caller was waiting in the queue, the
estimated hold time was announced as expected (estimated hold time is
less than 2 minutes ...).
Now the caller gets an announcement of their sequence in the queue
(Your call is now first in line ...).
I believe that the only changes I have made to queues.conf and
agents.conf is the addition of the context= statement and editing
the list of agents.
Has anyone else seen this? What am I missing?
regards,
Drew
QUEUES.CONF
[general]
persistentmembers = yes
[FxQueue]
music=default
strategy=rrmemory
context = opt-out_fxq
timeout=15
retry=3
wrapuptime=0
maxlen=0
announce-frequency=60
announce-holdtime = yes
reportholdtime=yes
memberdelay=1
servicelevel=120 ; seconds
member = Agent/1102
member = Agent/1103
member = Agent/1104
member = Agent/1105
AGENTS.CONF
[general]
persistentagents=yes
[agents]
ackcall=no
wrapuptime=0
musiconhold = default
agent = 1102,1234,Carly
agent = 1103,1234,Sean
agent = 1104,1234,Ed
agent = 1105,1234,Neil
EXTENSIONS.CONF (extract)
[call-centre]
exten = s,1,Noop(Entering Call Centre)
exten = s,n,Answer()
exten = s,n,Wait(1)
exten = s,n,Playback(welcome-fxqueue)
exten = s,n,Goto(5210,1)
; FxQueue
exten = 5210,1,Noop()
exten = 5210,n,Ringing()
exten = 5210,n,Wait(2)
exten = 5210,n,Queue(FxQueue|tH)
exten = 5210,n,Hangup()
exten = _6XXX,1,macro(ccexten,${EXT_${EXTEN}})
[macro-ccexten]
exten = s,1,Set(EXT=${ARG1})
exten = s,2,GotoIf([$:{EXT}]?4:3)
exten = s,3,Goto(i,1)
exten = s,4,Dial(${EXT},10,tT)
exten = i,1,Playback(pbx-invalid)
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[asterisk-users] Sender phone ringing while recipient talking
I've had asterisk running for about a month now between our PBX and our T1, and everything seems fine but for one simple nit-pick: When a call to the outside workd is made, and if the recipient picks up while a the sender's phone is still relaying the ring, the sender won't be heard until after the ring stops. This often translates a simple hello? into a lo? or even *long pause* hello, is anyone there? Is there a way to immediately stop the ring when a pickup is detected? Thanks, Nathan Bell ___ --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] transfers and CDR
Hi everybody, A question, how do I follow a call that is transferred? is the any event or something in the CDR that would let me find all the call sequence? Thanks Rodrigo ___ --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] Sender phone ringing while recipient talking
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nathan Bell Sent: Thursday, March 08, 2007 1:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sender phone ringing while recipient talking I've had asterisk running for about a month now between our PBX and our T1, and everything seems fine but for one simple nit-pick: When a call to the outside workd is made, and if the recipient picks up while a the sender's phone is still relaying the ring, the sender won't be heard until after the ring stops. This often translates a simple hello? into a lo? or even *long pause* hello, is anyone there? Is there a way to immediately stop the ring when a pickup is detected? Thanks, Nathan Bell How does your dial statement look in extensions.conf? Does it have the r option for your PSTN route? Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --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 distributed deployment
I just completed a deployment of 8 sites connected via MPLS, and I chose to go with the local * servers option and Sangoma hardware at each site. I then put dundi in place to route calls between sites and will later look at adding LCR. I'm with Steve on the cards, don't skimp on cards or even echo canceling. Most of my sited were 2-5 employees and I used Dell Optiplex systems for their servers, overkill on capabilities, but easy to maintain parts for. On 3/8/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like: Asterisk servers on all locations(central and remote offices) or Asterisk on Central office plus FXO Gateways on remote offices, all of this connected through a central asterisk cluster With the first option i have TDM cards seller that offer me DIGIUM (expensive) or OPENVOX (less expensive), but because i not have experience with OPENVOX telephony hardware I cant consider that. So, if Any can give me some good reasons for use OPENVOX against DIGIUM cards i would have solve this question because may build IAX trunks on each office. With the 2nd option I have sellers that offer me gateways: Quintum Tenor AFT400 Planet VIP-480 FO But, again, I don't have experience with asterisk and FXO gateways to think that it is the best solution amen that is the less expensive solution. Another solution that i consider is mount asterisk on central office and IP PBX DIGISTAR preconfigured on remote offices. On the Users Side I was considering the use of Ata's or FXS Gateways, with Ata's I get offers of Audiocodes MP202, GRANDSTREAM HT 386 or Linksys SPA-2002. And with FXS Gateways sellers offers me Quintum Tenor AXG2400, Quintum Tenor AFG800 Thanks for any word that can help me to get this VoIP deployment working and sorry for my english. Cheers G. ___ --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 -- Bruce Nortex Networks ___ --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] Number of groups?
Hi, I have an application with many outgoing analog ringdown trunks, 64 and was wondering is it better to make these all part of a single group (zapata.conf, group=), or give each one a different group, as they each go to a different place. If I give them each their own group so as to be able to refer to them as g0, g1, etc is there an upper limit on the number of groups? Thanks! --- Andrew___ --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] outdial to phone for new VM notification
Hi all, Does anyone have an application/script or extensions.conf file which will do the following? When a new VoiceMail is left for a user, the asterisk system will place a call to a cellphone/pstn number(via some provider). When the user answers his cell/home phone, comedian mail will ask for his password and he can check his Asterisk VM? Anyone have any examples of it working? If not, how hard would this be to implement. TIA! Cheers ___ --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] Empty Wildcard TDM400P as a MeetMe timer.
You don't need the power cable. It is only there to provide the necessary ring voltage to anything you may have plugged into installed _FXS_ modules. Henry Cobb wrote: I've just moved into 3.3v PCI servers and found that my clone X100P cards were lying about the 3.3v supported notch. Can I use a Wildcard TDM400P without any modules as a timer for MeetMe in a 64 bit 3.3v server? Will I still need to plug the hard disk power cable into it? Is there a better cheaper 3.3v MeetMe timer? (Boss doesn't trust the kernel timer.) -HJC ___ --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] Sender phone ringing while recipient talking
Are you using the option r in your Dial string? If so, remove it. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Bell Sent: Thursday, March 08, 2007 1:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sender phone ringing while recipient talking I've had asterisk running for about a month now between our PBX and our T1, and everything seems fine but for one simple nit-pick: When a call to the outside workd is made, and if the recipient picks up while a the sender's phone is still relaying the ring, the sender won't be heard until after the ring stops. This often translates a simple hello? into a lo? or even *long pause* hello, is anyone there? Is there a way to immediately stop the ring when a pickup is detected? Thanks, Nathan Bell ___ --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[2]: [asterisk-users] auto dialer
Not at all. :) I get myself confused with the same thing once in a while, cause the names are, to me at least, too similar. :) []'s MM -Original Message- From: Hall, Eric M. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Wed, 7 Mar 2007 17:08:22 -0500 Delivered: Wed, 07 Mar 2007 19:06:08 Subject:[asterisk-users] auto dialer OK now I fell like a a$$... Thanks for that kick in the butt !! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Melcon Moraes Sent: Wednesday, March 07, 2007 4:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] auto dialer WaitTime stands for how long to wait until the call is considered NO ANSWERED Who can pickup a phone in 2 seconds, if not a robot? Try switch values between Retrytime and WaitTime. []'s MM -Original Message- From: Hall, Eric M. [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: Sent: Wed, 7 Mar 2007 15:53:23 -0500 Delivered: Wed, 07 Mar 2007 17:45:35 Subject:[asterisk-users] auto dialer Not able to get the auto dialer part of asterisk to workwith the zap channel. It works great with the sip channel. Here is the callfile and the CLI output Call File Channel: ZAP/G1/6144994925 MaxRetries: 3 RetryTime: 40 WaitTime: 2 Context: amaxx Extension: 36652 Priority: 1 CLI Output Connected to Asterisk SVN-branch-1.4-r57207 currentlyrunning on VoIP-PBX (pid = 8002) Verbosity is at least 3 -- Attempting call on ZAP/G1/6144994925 [EMAIL PROTECTED]:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- Hungup 'Zap/23-1' [Mar 7 15:46:29] NOTICE[10159]: pbx_spool.c:341attempt_thread: Call failed to go through, reason 0 VoIP-PBX*CLI E-mail classificado pelo Identificador de Spam Inteligente. Para alterar a categoria classificada, visite o http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l= 1,1173300915.746475.15282.aldavila.hst.terra.com.br,8031,Des15,Des15Ter ra Mail --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --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 E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1173305169.758007.26033.baladonia.hst.terra.com.br,5525,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --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] Accessing Voicemail by dialing own number
I searched google for asterisk voicemail documentation and could not find anything. After more searching, I found someone who had done it. If you create an a extension in the current context, it will be called when someone presses the asterisk during the outgoing message. -- Chris Carey On 3/8/07, Andrew Joakimsen [EMAIL PROTECTED] wrote: Yes, you can setup * key to do that, its a standard feature see the docs of the voicemail application for details on how to do it. ___ --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] Packet2Packet Bridging Questions
OK...that makes much more sense. So here's my follow-up question: what's the easiest way to check if I'm native bridging a call. I'm trying to offload as much RTP traffic as possible, and want to have a way to check quickly (there are well over 50 calls on each of these boxes at any given time). I've been going the ethereal route, which is great for debugging, but not so good for a quick look. Thanks again, Daryl ___ --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] Empty Wildcard TDM400P as a MeetMe timer.
The trick is modifying the source in zaptel file: wctdm.c and changing to the following then doing a make clean, make make install. static int timingonly = 1; The original value was a zero. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Thursday, March 08, 2007 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer. You don't need the power cable. It is only there to provide the necessary ring voltage to anything you may have plugged into installed _FXS_ modules. Henry Cobb wrote: I've just moved into 3.3v PCI servers and found that my clone X100P cards were lying about the 3.3v supported notch. Can I use a Wildcard TDM400P without any modules as a timer for MeetMe in a 64 bit 3.3v server? Will I still need to plug the hard disk power cable into it? Is there a better cheaper 3.3v MeetMe timer? (Boss doesn't trust the kernel timer.) -HJC ___ ___ --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] Coaching in asterisk
Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx attachment: winmail.dat___ --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: Asterisk distributed deployment
I couldn't agree more. The Telco card is the LAST thing you should be trying to cut corners on. IMHO you should consider a Sangoma A200D which is even more money due to the HWEC. It's worth every penny! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, March 08, 2007 12:36 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisk distributed deployment Steve, Im not asking but looking for a suggest about multiple solutions to the same problem, Im looking for experinces with hibrid deployments that save me money, for example sellers offers me TDM04B DIGIUM CARDS about u$s 500 against u$s 150 for OPENVOX CARDS. Cheers I think you will pay in the long run if you are going to skimp on U$D350. Couldn't one dropped call from a prospective big time customer be worth more than U$D350? Would spending a week of time trying to get things working correctly be worth U$D350? I would suggest doing it right and not skimping. Your phone system should be transparent and is a direct reflection on your business. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --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: Coaching in asterisk
Yep, it's called Whisper Check in voip-info.org I think I've read stuff about it there. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Thursday, 8 March 2007 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx ___ --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] No application 'Prefix' for extension in1.2x, what app I have to use instead?
Hi I want to use Prefix app in extensions but get this error: WARNING[9255] pbx.c: No application 'Prefix' for extension ... I am just want to do somethig like this: exten = _9XXX,1,ANSWER() exten = _9XXX,2,Wait(1) exten = _9XXX,3,Prefix(511) exten = _5119XXX,4,DeadAGI(a2billtest.php|1) exten = _5119XXX,5,Hangup() Please someone tell me how to install Prefix/Suffix application, or tell if it has been deprecated in 1.2.x versions... what command I have to use instead? ___ --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 SIP to MAX TNT Gateway, Sporadic Echo
Hi All, I'm trying to track down an intermittent echo issue. My setup is phonesipasterisksiptntpri to carrier less than 10ms latency on the network, 100% SIP, ULAW I have several different phones; cisco, linksys, polycom, snom. It's difficult for me to reproduce the problem regularly so I'm really having trouble isolating anything. I'm wondering if this could be a bad DSP on the TNT, and how would I isolate. We have 600+ DSP's in this chassis. Any experience or ideas with this type of issue would greatly be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ --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: Coaching in asterisk
You must be talking about Chanspy. It is included in 1.4. Has anyone tried to compiled for 1.2x? -Original Message- From: [EMAIL PROTECTED] on behalf of Dean Collins Sent: Thu 3/8/2007 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RE: Coaching in asterisk Yep, it's called Whisper Check in voip-info.org I think I've read stuff about it there. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Thursday, 8 March 2007 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx ___ --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 attachment: winmail.dat___ --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] Zap Channel Deadlocks
Hey List, Asterisk 1.2.13 with Sangoma Card and beta 14 drivers. I am having problems with deadlock channels and having to kill asterisk, and then restart it, cannot make calls in or outbound. This has happend about 4 times now, and the system was running fine for a few months fine. Any suggestions or comments would be greaet, and im in a world of hurt here! Thanks Brad ___ --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] Call load balancing
I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --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] Coaching in asterisk
Yes. I believe its called whisper mode. Have a look on voip-info.org - Original Message - From: Wai Wu [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 08, 2007 11:25 PM Subject: [asterisk-users] Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx ___ --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 application 'Prefix' for extension in1.2x, what app I have to use instead?
Rafael J. Risco G.V. wrote:
Hi
I want to use Prefix app in extensions but get this error:
WARNING[9255] pbx.c: No application 'Prefix' for extension ...
I am just want to do somethig like this:
exten = _9XXX,1,ANSWER()
exten = _9XXX,2,Wait(1)
exten = _9XXX,3,Prefix(511)
exten = _5119XXX,4,DeadAGI(a2billtest.php|1)
exten = _5119XXX,5,Hangup()
Please someone tell me how to install Prefix/Suffix application, or
tell if it has been deprecated in 1.2.x versions... what command I
have to use instead?
Prefix was deprecated in 1.0 and removed in 1.2
Use this instead of Prefix():
exten = _9XXX,1,ANSWER()
exten = _9XXX,2,Wait(1)
exten = _9XXX,3,Goto(511${EXTEN},1)
exten = _5119XXX,1,DeadAGI(a2billtest.php|1)
exten = _5119XXX,2,Hangup()
___
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Re: [asterisk-users] RE: Coaching in asterisk
There's a lot more than just app_chanspy.c changes required to get the full functionality backported to 1.2. On 3/8/07, Wai Wu [EMAIL PROTECTED] wrote: You must be talking about Chanspy. It is included in 1.4. Has anyone tried to compiled for 1.2x? -Original Message- From: [EMAIL PROTECTED] on behalf of Dean Collins Sent: Thu 3/8/2007 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RE: Coaching in asterisk Yep, it's called Whisper Check in voip-info.org I think I've read stuff about it there. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Thursday, 8 March 2007 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx ___ --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 -- 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
[asterisk-users] Re: Coaching in asterisk
Wai Wu wrote: Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx I think whisper coaching is implemented in version 1.4. Thanks, Steve ___ --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: Coaching in asterisk
Ouch, I just have to move to 1.4. Is 1.4 stable at all under heavy load? -Original Message- From: [EMAIL PROTECTED] on behalf of BJ Weschke Sent: Thu 3/8/2007 5:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Coaching in asterisk There's a lot more than just app_chanspy.c changes required to get the full functionality backported to 1.2. On 3/8/07, Wai Wu [EMAIL PROTECTED] wrote: You must be talking about Chanspy. It is included in 1.4. Has anyone tried to compiled for 1.2x? -Original Message- From: [EMAIL PROTECTED] on behalf of Dean Collins Sent: Thu 3/8/2007 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RE: Coaching in asterisk Yep, it's called Whisper Check in voip-info.org I think I've read stuff about it there. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Thursday, 8 March 2007 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx ___ --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 -- 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 attachment: winmail.dat___ --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] Sender phone ringing while recipient talking
Thanks, that fixed the problem. I didn't realise that the 'r' wasn't necessary to signal the ring to the sender. Bill Gibbs wrote: Are you using the option r in your Dial string? If so, remove it. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Bell Sent: Thursday, March 08, 2007 1:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sender phone ringing while recipient talking I've had asterisk running for about a month now between our PBX and our T1, and everything seems fine but for one simple nit-pick: When a call to the outside workd is made, and if the recipient picks up while a the sender's phone is still relaying the ring, the sender won't be heard until after the ring stops. This often translates a simple hello? into a lo? or even *long pause* hello, is anyone there? Is there a way to immediately stop the ring when a pickup is detected? Thanks, Nathan Bell ___ --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] 1.4 compile issue
I am use Fedora 3, and run into a 1.4 compile issue. When 'make install' I got this message. [EMAIL PROTECTED] asterisk-1.4.1]# make install make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list-next != 0' failed. make: *** [utils] Aborted [EMAIL PROTECTED] asterisk-1.4.1]# ___ --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: Coaching in asterisk
NVWhisper. Justin -- Date: Thu, 08 Mar 2007 16:25:28 -0500 From: Wai Wu [EMAIL PROTECTED] Subject: [asterisk-users] Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx TV dinner still cooling? Check out Tonight's Picks on Yahoo! TV. http://tv.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
RE: [asterisk-users] Sender phone ringing while recipient talking
It creates an artificial ring and can be helpful when the telco or carrier does not provide ringing (which they should). Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nathan Bell Sent: Thursday, March 08, 2007 5:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sender phone ringing while recipient talking Thanks, that fixed the problem. I didn't realise that the 'r' wasn't necessary to signal the ring to the sender. Bill Gibbs wrote: Are you using the option r in your Dial string? If so, remove it. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Bell Sent: Thursday, March 08, 2007 1:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sender phone ringing while recipient talking I've had asterisk running for about a month now between our PBX and our T1, and everything seems fine but for one simple nit-pick: When a call to the outside workd is made, and if the recipient picks up while a the sender's phone is still relaying the ring, the sender won't be heard until after the ring stops. This often translates a simple hello? into a lo? or even *long pause* hello, is anyone there? Is there a way to immediately stop the ring when a pickup is detected? Thanks, Nathan Bell ___ --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] OT Vonage V-Phone Adapter (Possible Hack)
Steve, If you can get this to work with your own choice of softphone please post back to the list. I've wondered about it myself. On 3/7/07, Steve Totaro [EMAIL PROTECTED] wrote: It would be cool to get one of these and see if it can be hacked and loaded with your favorite SIP or IAX softphone. Looking at the pic, it looks like the dongle is both a soundcard and memory stick. Heck, I would be glad to have it if I could get the soundcard to work. Might as well since it is free after rebate. http://www.circuitcity.com/ssm/Accessories-for-Vonage-V-Phone-VPHONE/sem /rpsm/oid/162059/rpem/ccd/productDetailAccessory.do#tabs Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --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 -- Bruce Nortex Networks ___ --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] Newbie Question
Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is not my issue (I think). Here is some info: 1. * server and clients are all on the same subnet but are separated from the internet by a proxy/firewall which forces all port 80 traffic through the proxy. 2. The server has a single channel fxo card. 3. Snip of sip.conf: [test] type=friend secret=verysecret regexten=1234 ; When they register, create extension 1234 callerid=Test Unit 1234 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw [EMAIL PROTECTED]; Subscribe to status of multiple mailboxes context=internal Here is the problem: Xlite registers fine. When I dial 500 to access the demo, the * console shows the client connect and the demo audio plays. However, there is no sound on the client end. I have installed Xlite on an XP workstation and on a *nix workstation. Both installs behave the same. Any thoughts? Or do I need to post more details? Thanks, Chris ___ --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] Call load balancing
On Thu, 8 Mar 2007, David Ruggles wrote: I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Quick answer, yes. How is more interesting :) First, unless your AGI's are massive or incredibly inefficient, 2 PRI's won't swamp your IVR boxes. I have 3 1u servers each with 2 PRI's forwarding all 138 calls to a single application server. All of the PRI's could be handled by 1 1u but management wanted flexibility and redundancy. The application server does IVR, conferencing, records messages, plays canned stories, credit card processing, etc, etc, etc. All implemented with a bunch of AGI's written in C. Each call executes a minimum of 9 AGI's and yes, some AGI consolidation is planned. All database work is handled by a separate box. Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] OT Vonage V-Phone Adapter (Possible Hack)
The real question is what quality softphones can run as an executable or without having to install anything? I assume that the Vonage softphone operates this way (can anyone confirm?) I am thinking about machines that are locked down. I guess the sound card will not install either in that case (but if it has an internal sound card, it should work). Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Thursday, March 08, 2007 6:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack) Steve, If you can get this to work with your own choice of softphone please post back to the list. I've wondered about it myself. On 3/7/07, Steve Totaro [EMAIL PROTECTED] wrote: It would be cool to get one of these and see if it can be hacked and loaded with your favorite SIP or IAX softphone. Looking at the pic, it looks like the dongle is both a soundcard and memory stick. Heck, I would be glad to have it if I could get the soundcard to work. Might as well since it is free after rebate. http://www.circuitcity.com/ssm/Accessories-for-Vonage-V-Phone-VPHONE/sem /rpsm/oid/162059/rpem/ccd/productDetailAccessory.do#tabs Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --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 -- Bruce Nortex Networks ___ --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] outdial to phone for new VM notification
While not what you are specifically requesting, making a call after a voicemail is left is covered at http://opensourcemadness.blogspot.com/2007/03/propagating-asterisk-mwi-a cross.html Using those techniques you can setup what you are describing. Rather than calling another Asterisk server, just have it call the user's phone number and connect it to VoiceMailMain. - Jeremy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of end1r Sent: Thursday, March 08, 2007 12:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] outdial to phone for new VM notification Hi all, Does anyone have an application/script or extensions.conf file which will do the following? When a new VoiceMail is left for a user, the asterisk system will place a call to a cellphone/pstn number(via some provider). When the user answers his cell/home phone, comedian mail will ask for his password and he can check his Asterisk VM? Anyone have any examples of it working? If not, how hard would this be to implement. TIA! Cheers ___ --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] 1.4 compile issue
Wai Wu wrote: I am use Fedora 3, and run into a 1.4 compile issue. When 'make install' I got this message. You need to update to a newer version of gnu make. -- Russell Bryant Software Engineer Digium, Inc. ___ --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] Newbie Question
If both the asterisk server and the softphone are on the same LAN then I would look at your firewall settings on the box. Make sure you have 5060 and 10,000 - 20,000 UDP open. If the phone is connecting to the server over the internet and the server IS behind NAT then you need to forward ports 5060 and 10,000-20,000 UDP to the asterisk server. - Original Message - From: Chris Nighswonger [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, March 09, 2007 1:16 AM Subject: [asterisk-users] Newbie Question Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is not my issue (I think). Here is some info: 1. * server and clients are all on the same subnet but are separated from the internet by a proxy/firewall which forces all port 80 traffic through the proxy. 2. The server has a single channel fxo card. 3. Snip of sip.conf: [test] type=friend secret=verysecret regexten=1234 ; When they register, create extension 1234 callerid=Test Unit 1234 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw [EMAIL PROTECTED]; Subscribe to status of multiple mailboxes context=internal Here is the problem: Xlite registers fine. When I dial 500 to access the demo, the * console shows the client connect and the demo audio plays. However, there is no sound on the client end. I have installed Xlite on an XP workstation and on a *nix workstation. Both installs behave the same. Any thoughts? Or do I need to post more details? Thanks, Chris ___ --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] Is Allison going to be banned from foreign travel over polar bears?
I read this story and thought of Allison's prompt to try not to think about blue eyed polar bears. Will she be banned from foreign travel now? Steve Prior -- snip -- WASHINGTON (Reuters) - Polar bears, sea ice and global warming are taboo subjects, at least in public, for some U.S. scientists attending meetings abroad, environmental groups and a top federal wildlife official said on Thursday. http://today.reuters.com/news/articlenews.aspx?type=topNewsstoryid=2007-03-08T222736Z_01_N08259521_RTRUKOC_0_US-POLARBEARS-SCIENTISTS.xmlsrc=rss ___ --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] 1.4 compile issue
Found out I need make version 3.8 or later -Original Message- From: [EMAIL PROTECTED] on behalf of Wai Wu Sent: Thu 3/8/2007 5:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 1.4 compile issue I am use Fedora 3, and run into a 1.4 compile issue. When 'make install' I got this message. [EMAIL PROTECTED] asterisk-1.4.1]# make install make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list-next != 0' failed. make: *** [utils] Aborted [EMAIL PROTECTED] asterisk-1.4.1]# ___ --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 attachment: winmail.dat___ --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] Zap Channel Deadlocks
Ummm. How about upgrading to production released drivers? -Original Message- From: Ron McCarthy [mailto:[EMAIL PROTECTED] Sent: Thursday, March 08, 2007 1:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zap Channel Deadlocks Hey List, Asterisk 1.2.13 with Sangoma Card and beta 14 drivers. I am having problems with deadlock channels and having to kill asterisk, and then restart it, cannot make calls in or outbound. This has happend about 4 times now, and the system was running fine for a few months fine. Any suggestions or comments would be greaet, and im in a world of hurt here! Thanks Brad ___ --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] Newbie Question
Don't forget about 4569 UDP port (IAX protocol) forwarded to your Asterisk box. Best Regards; Leonardo Kamache On 3/8/07, Dovid B [EMAIL PROTECTED] wrote: If both the asterisk server and the softphone are on the same LAN then I would look at your firewall settings on the box. Make sure you have 5060 and 10,000 - 20,000 UDP open. If the phone is connecting to the server over the internet and the server IS behind NAT then you need to forward ports 5060 and 10,000-20,000 UDP to the asterisk server. - Original Message - From: Chris Nighswonger [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, March 09, 2007 1:16 AM Subject: [asterisk-users] Newbie Question Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is not my issue (I think). Here is some info: 1. * server and clients are all on the same subnet but are separated from the internet by a proxy/firewall which forces all port 80 traffic through the proxy. 2. The server has a single channel fxo card. 3. Snip of sip.conf: [test] type=friend secret=verysecret regexten=1234 ; When they register, create extension 1234 callerid=Test Unit 1234 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw [EMAIL PROTECTED]; Subscribe to status of multiple mailboxes context=internal Here is the problem: Xlite registers fine. When I dial 500 to access the demo, the * console shows the client connect and the demo audio plays. However, there is no sound on the client end. I have installed Xlite on an XP workstation and on a *nix workstation. Both installs behave the same. Any thoughts? Or do I need to post more details? Thanks, Chris ___ --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] Newbie Question
Thanks for the responses. iptables on the * box has no rules and all tables default to 'accept.' I have not got to the point of placing calls out across the internet yet. The issue here is no audio back from the * box when running through the demo routine. I'll try to set it up to make a call outside tomorrow. Chris On 3/8/07, Leonardo Kamache (Gmail) [EMAIL PROTECTED] wrote: Don't forget about 4569 UDP port (IAX protocol) forwarded to your Asterisk box. Best Regards; Leonardo Kamache On 3/8/07, Dovid B [EMAIL PROTECTED] wrote: If both the asterisk server and the softphone are on the same LAN then I would look at your firewall settings on the box. Make sure you have 5060 and 10,000 - 20,000 UDP open. If the phone is connecting to the server over the internet and the server IS behind NAT then you need to forward ports 5060 and 10,000-20,000 UDP to the asterisk server. - Original Message - From: Chris Nighswonger [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, March 09, 2007 1:16 AM Subject: [asterisk-users] Newbie Question Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is not my issue (I think). Here is some info: 1. * server and clients are all on the same subnet but are separated from the internet by a proxy/firewall which forces all port 80 traffic through the proxy. 2. The server has a single channel fxo card. 3. Snip of sip.conf: [test] type=friend secret=verysecret regexten=1234 ; When they register, create extension 1234 callerid=Test Unit 1234 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw [EMAIL PROTECTED]; Subscribe to status of multiple mailboxes context=internal Here is the problem: Xlite registers fine. When I dial 500 to access the demo, the * console shows the client connect and the demo audio plays. However, there is no sound on the client end. I have installed Xlite on an XP workstation and on a *nix workstation. Both installs behave the same. Any thoughts? Or do I need to post more details? Thanks, Chris ___ --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 -- Chris Nighswonger Network Systems Director Foundations Bible College Seminary www.foundations.edu www.fbcradio.org [EMAIL PROTECTED] V:910-892-8761 C:919-820-5473 - NOTICE: The information contained in this electronic mail message is intended only for the use of the intended recipient, and may also be protected by the Electronic Communications Privacy Act, 18 USC Sections 2510-2521. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please reply to the sender, and delete the original message. Thank you. ___ --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] Is Allison going to be banned from foreign travel over polar bears?
Steve Prior wrote: I read this story and thought of Allison's prompt to try not to think about blue eyed polar bears. Will she be banned from foreign travel now? I supposed it's ok since blue-eyed polar bears are fictitious and thus protected by the first amendment :) Leo ___ --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] Boot order of 2 TE110P and 1 TDM400P in the same machine
Hello Everyone, I checked with zttool that sometimes after the machine boots the order of the boards is changed like this: â Alarms Span â OK Digium Wildcard TE110P T1/E1 Card 0 â OK Digium Wildcard TE110P T1/E1 Card 1 â OK Wildcard TDM400P REV I Board 1 and sometimes: â Alarms Span â OK Wildcard TDM400P REV I Board 1â OK Digium Wildcard TE110P T1/E1 Card 1 â OK Digium Wildcard TE110P T1/E1 Card 0 What do I have to configure in order to the boards appear in the same position and the configuration work always?? best regards, Pablo - Get your own web address. Have a HUGE year through Yahoo! Small Business.___ --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] Zap Channel Deadlocks
I gues ill look and see what version they are on, its a production system, so that always scares me!!! But, good ideal!! :) On 3/8/07, shadowym [EMAIL PROTECTED] wrote: Ummm. How about upgrading to production released drivers? -Original Message- From: Ron McCarthy [mailto:[EMAIL PROTECTED] Sent: Thursday, March 08, 2007 1:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zap Channel Deadlocks Hey List, Asterisk 1.2.13 with Sangoma Card and beta 14 drivers. I am having problems with deadlock channels and having to kill asterisk, and then restart it, cannot make calls in or outbound. This has happend about 4 times now, and the system was running fine for a few months fine. Any suggestions or comments would be greaet, and im in a world of hurt here! Thanks Brad ___ --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: Coaching in asterisk
We used ChanSpy to allow a supervisor to listen in on the calls of their staff. There was one huge problem with this, which I imagine would affect whisper as well. The supervisor typically sat fairly close to the worker, and could hear both the voice of the worker as they spoke AND the delayed voice coming through their head phones. It was rather distracting and made it difficult to really be practical. Doug. Dean Collins wrote: Yep, it's called Whisper Check in voip-info.org I think I've read stuff about it there. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Thursday, 8 March 2007 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx ___ --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: Coaching in asterisk
Must be a quiet and small call center without high cubicle walls. There is no way that would be an issue at the call center I setup. 16 agents to a team and all of them on the phone all the time, you cannot even fix in on an agent if you wanted to, there was too much noise. Thanks, Steve Doug Garstang wrote: We used ChanSpy to allow a supervisor to listen in on the calls of their staff. There was one huge problem with this, which I imagine would affect whisper as well. The supervisor typically sat fairly close to the worker, and could hear both the voice of the worker as they spoke AND the delayed voice coming through their head phones. It was rather distracting and made it difficult to really be practical. Doug. Dean Collins wrote: Yep, it's called Whisper Check in voip-info.org I think I've read stuff about it there. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Thursday, 8 March 2007 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx ___ --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] Issues with a Linksys SPA 2102 and asterisk
Topology: analog_phone-SPA2102-Navini_Wireless_Router--ISP--Asterisk A ping against the asterisk server shows aprox 145ms roundtrip. 128kbps upstream 512kbps downstream g729a as codec signal quality of the navini router: 100% The ATA operates correctly in every form, however sometimes when someone is talking to me (the other person is at pstn) and then I start talking the other end receives garbled voice and i need to start talking again. So I played with the jitter buffers in the available modes (low, medium, high) (direction upward, downward both) and it seems i cannot improve my voice experience. Any gurus out there with experience with the SPA 2102 against asterisk 1.2.14? thanks, -- Erick Perez ___ --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: Help: CallerID Name not being sent on outbound PRI trunk
Tomislav, really? and how does it show up on my POTS line? On 3/8/07, Tomislav Parcina [EMAIL PROTECTED] wrote: Matt wrote: Thanks I was just about to say this. You CAN'T send caller-id-name. To be able to set name you need to set it with Telcordia or whomever manages numbers in your country. Optima provider in Croatia allows users to set up CallerID name on outgoing PRI calls. -- Tomislav Parcina [EMAIL PROTECTED] ___ --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: Help: CallerID Name not being sent on outbound PRI trunk
This means your POTS provider's hardware is not blocking CNAM which is very strange, and if they would find out people are using Asterisk to send custom CNAM values on their system, they'll block it immediately. PRI provider can also open passage to custom CNAM, but no one does it. On 3/9/07, C F [EMAIL PROTECTED] wrote: Tomislav, really? and how does it show up on my POTS line? On 3/8/07, Tomislav Parcina [EMAIL PROTECTED] wrote: Matt wrote: Thanks I was just about to say this. You CAN'T send caller-id-name. To be able to set name you need to set it with Telcordia or whomever manages numbers in your country. Optima provider in Croatia allows users to set up CallerID name on outgoing PRI calls. -- Tomislav Parcina [EMAIL PROTECTED] ___ --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 -- Zeeshan A Zakaria ___ --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] Which VoIP router and switch to use for medium size business
Hi everybody, What is a proper setup for a medium size business with about 20 IP phones and 20 computers. Right now they are using a regular Linksys router which we use at homes. Their switch is also a very standard switch. Now they need to put there something better and VoIP compatible. What people use out there in serious and professional VoIP installations for medium size businesses? Is there a good 24 port router with VoIP compatibility with no need of an extra switch? Please advice me for all the equipment I'd need for a complete network upgrade. Thanks -- Zeeshan A Zakaria ___ --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] Fwd: Can't hear any sound
Note: forwarded message attached. Send instant messages to your online friends http://uk.messenger.yahoo.com ---BeginMessage--- Hey, I am new to asterisk and softphones. I am able to install astersik and 2 XLite softphones on three PCs with linux feora core 6. I have also written a basic dial plan to make calls between two clients.But when i dial from a pc to another PC the calls goes through i can hear the ring tone and also recieve call but i can't hear any voice. Please could anyone help me. Regards, Szabstians Send instant messages to your online friends http://uk.messenger.yahoo.com ---End 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
[asterisk-users] Zaptel problem after upgrading to 1.2.16
Hi guys, I'm hoping I've made a silly mistake here, but I've been staring at the screen for the past few hours and I can't work it out. I upgraded to 1.2.16 recently, and am having problems with zaptel. The card is detected, I get a reasonable output from ztcfg -vv, and zttool shows the installed module (TDM400) with one FXS module. But when I start asterisk, I get an error saying that my IAX connection won't work in trunked mode because there's no timing interface. Zaptel doesn't show up in the output of show channeltypes. Should there be a problem with using the trunk version of zaptel, but 1.2.16 of asterisk? Are there any places that I can specifically load/enable the zaptel module? Any help much appreciated before I go insane... J Regards, Mark. ___ --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
