Re: [asterisk-users] Want new standard Asterisk prompts? Just ask!
On Friday 08 June 2012, Steve Sokol wrote: Allison Smith has kindly agreed to add some new prompts to the additional prompt set for Asterisk. If you have ideas for additional stock prompts (serious or silly), please submit them: https://docs.google.com/spreadsheet/viewform?formkey=dERsMVdGNjBwVmVGWlUzcm t3ZzJTMFE6MQ She will record the prompts on an occasional basis and we will get them set up for download. Many of the default prompts talk about the pound key. But British phones do not have a £ sign on the keypad; instead, we have a comment mark (#) key to the right of the zero, usually called hash. So, some stock prompts mentioning the hash key instead of the pound key would be nice. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk on VMware ESX
Blake Burgess wrote: We had a bunch of voice quaity issues which took ages to diagnose because of this. Obviously if you have a DAHDI card that your passing through to the vm or one of thesehttp://wiki.sangoma.com/sangoma-wanpipe-voicetime you can avoid this And, trying this last week with a Digium T110P kept failing with HDLC errors on D channel. Tried it under ESXi 5. Ended up moving to a real box. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Want new standard Asterisk prompts? Just ask!
A J Stiles wrote: But British phones do not have a £ sign on the keypad; instead, we have a comment mark (#) key to the right of the zero, usually called hash. Just a note, the hash key in the US is called the pound key. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting span orders with Astribank and Sangoma A101
On Sun, Jun 10, 2012 at 10:10:29PM +1200, Duncan Turnbull wrote: Hi All Just a quick check on the best way to ensure multiple cards/devices load in the correct order. Asterisk 1.8 with Sangoma A101 had no problems until we introduced an Astribank. root@pabx377:/etc/asterisk# dahdi_hardware -v usb:001/004 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware LABEL=[usb:X1060395] CONNECTOR=@usb-:00:1d.7-3 XBUS-00/XPD-00: FXS (14) Span 2 XBUS-00/XPD-10: FXS (8) Span 3 XBUS-00/XPD-20: FXS (8) Span 4 XBUS-00/XPD-30: FXS (8) Span 5 pci::05:00.0 wanpipe- 1923:0300 Sangoma Technologies Corp. A101 single-port T1/E1 But now they have changed order after reboot to 1-4 and I am guessing that makes the A101 span 5 but I want to make these setting permanent I see mention of /etc/dahdi/xpp_order and dahdi_genconf xpporder but will that help me set the spans within Dahdi so they always appear on the same number The word load is not the correct one here. The Astribank's span will only appear once you run 'dahdi_registration on' . Normally this happens in the dahdi init script, but you can tweak it to happen elsewhere. Another option: if Dahdi 2.6 is an option (no idea if Sangoma supports it): that version has an option to reserve span numbers explicitly for specific hardware devices. With this we can do away with relying on the load order. This is considered experimental (and I wonder how its scriptary interacts with the Sangoma scriptary). http://docs.tzafrir.org.il/dahdi-linux/#_span_assignments , but there are better ways. Let me know what you want to do. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk on VMware ESX
It seems like much of the problem with virtualizing Asterisk is getting it to interface with DAHDI cards. Is that correct? As I'm not planning on using such cards (I'm only interfacing with our broadband connection), would this make virtualizing more feasible? Richard -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Monday, June 11, 2012 4:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Asterisk on VMware ESX Blake Burgess wrote: We had a bunch of voice quaity issues which took ages to diagnose because of this. Obviously if you have a DAHDI card that your passing through to the vm or one of thesehttp://wiki.sangoma.com/sangoma-wanpipe-voicetime you can avoid this And, trying this last week with a Digium T110P kept failing with HDLC errors on D channel. Tried it under ESXi 5. Ended up moving to a real box. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk on VMware ESX
Hiers, Richard wrote: It seems like much of the problem with virtualizing Asterisk is getting it to interface with DAHDI cards. Is that correct? As I'm not planning on using such cards (I'm only interfacing with our broadband connection), would this make virtualizing more feasible? I'd like to say yes, but since I failed with my Digium card, I've moved on to a real box. I was planning on a week worth of testing on ESXi 5, if it had been successful with PCI Passthough. On the other hand, virtualizing my MythTV backup with a TV Card with PCI Passthough, works a treat! Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk on VMware ESX
We virtualize every asterisk install, and have achieved density levels of 80MB RAM per install of asterisk. We do it all day, every day. As Chris wrote if you're putting it on shared hardware that you don't control, just don't. If you control all of the hardware it's very doable. Thanks David On Sun, Jun 10, 2012 at 3:07 PM, Chris Bagnall aster...@lists.minotaur.ccwrote: On 8/6/12 9:17 pm, Hiers, Richard wrote: I don't expect to need to use any special hardware, just a sip trunk over our broadband connection. We have about 150 phones at present. Is ESX a viable platform for us? And second, what is the recommended virtual configuration (mem, cpu, etc.)? Any other considerations? I think the concern expressed about running Asterisk on a virtualised platform is more to do with the impact the other load on the host machine might have on your Asterisk VM. If you're using ESX in a shared hosting environment where you have very little control over the other workload on that host, then sooner or later there's a risk your VM is going to experience spikes in latency. On the other hand, if you're running a virtualised platform internally where you can control precisely the load on the host machine, then you'll probably find you're fine. FWIW, we run Asterisk under Xen in production. Some of the VMs have well over a thousand connected SIP devices and we've yet to encounter significant problems. But we're able to control the other VMs on the hosts very precisely: the only other VMs running on those hosts provide low-load services such as rsync for remote backup (which is only used late at night when call load is low on the Asterisk VMs). Running Asterisk in a VM, even if it's the only VM on that host, does give you some considerable benefits in the event of host machine failover: hardware independence and live migration are the two that spring immediately to mind. Kind regards, Chris -- This email is made from 100% recycled electrons -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- -- www.ringfree.biz 828-575-0030 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Get voicemail box password from dialplan?
I would like to be able to use the dialing extension's voicemail box password
to authenticate or as a PIN code in the dialplan. Is there a best method for
doing this? I could use AGI scripting but I was hoping there was a built-in
dialplan means for
doing this. I have used VMAuthenticate but I would like more flexibility than
what this offers.
Also, related to this question; is there a best or recommended method to
determine the dialing extensions voice mail box? I have been using variations
of ${CUT(CHANNEL,-,1)} which does work but I just have to be aware of how I
name my devices.
Thank you for your help.
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Re: [asterisk-users] Get voicemail box password from dialplan?
On Mon, Jun 11, 2012 at 8:34 AM, Chet W. Stevens
cwstev...@interact.ccsd.net wrote:
Also, related to this question; is there a best or recommended method to
determine the dialing extensions voice mail box? I have been using
variations of ${CUT(CHANNEL,-,1)} which does work but I just have to be
aware of how I name my devices.
I use the following method to get the mailbox for a peer under 1.4:
exten = *98,n,Set(peer=${SIPCHANINFO(peername)}) ; Get the peer
exten = *98,n,Set(mailbox=${SIPPEER(${peer},mailbox)}); Get the mailbox
I dont know about it being the best or recommended method but I've yet
to have it break on me.
--
A human being should be able to change a diaper, plan an invasion,
butcher a hog, conn a ship, design a building, write a sonnet, balance
accounts, build a wall, set a bone, comfort the dying, take orders,
give orders, cooperate, act alone, solve equations, analyze a new
problem, pitch manure, program a computer, cook a tasty meal, fight
efficiently, die gallantly. Specialization is for insects.
---Heinlein
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[asterisk-users] Digium IP Phones D40
Hi All; Any one used Digium IP Phones D40? I need to know if they are stable with good voice quality? Comparing to Polycom 330, which is better? Let us talk frankly although I know that we have to support Digium. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium IP Phones D40
On Mon, Jun 11, 2012 at 9:58 AM, bilal ghayyad bilmar...@yahoo.com wrote: I need to know if they are stable with good voice quality? Comparing to Polycom 330, which is better? Let us talk frankly although I know that we have to support Digium. Voice quality is great. I would choose the Digium phones over a Polycom every time, that's an easy choice. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Differences between PBX and SBC
Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium IP Phones D40
On 12-06-11 12:58 PM, bilal ghayyad wrote: Hi All; Any one used Digium IP Phones D40? I need to know if they are stable with good voice quality? Comparing to Polycom 330, which is better? Let us talk frankly although I know that we have to support Digium. I don't think these topics about comparing A to B work very well. For me, it comes down to what has worked well in the past. With that in mind, it will be hard for us to give up Polycom phones. I had the ability to test the Digium phones while at Digium, and they are rugged phones, they look professional too. However, I only tested with them for a month or so. Now the Digium phones have a tighter integration out of the box with Asterisk / Switchvox however that is not an added feature since we already have provisioning modules for Polycom phones. If you have never mass deployed Polycom phones, it does require some work. You need to get your hands dirty but Polycom has a lot of documentation about the process. With Digium, they take this point of pain away from you. A ship the phones with a tight integration with asterisk / Switchvox. There are some other features about visual voicemail and JS applications, however I don't require them so not a feature I am interested in. So, to answer your questions, compare Polycom to Digium. For me, the winner is Polycom because their phones have been around for years. Digium's only months. And because we have Polycom phones at 95% of the sites we manage, adding another vendor into the mix for use to support does not make sense at this time. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between PBX and SBC
On 12-06-11 02:06 PM, Danny Dias wrote: Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment What are you expecting the SBC to do? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get voicemail box password from dialplan?
On 06/11/2012 10:34 AM, Chet W. Stevens wrote: I would like to be able to use the dialing extension's voicemail box password to authenticate or as a PIN code in the dialplan. Is there a best method for doing this? I could use AGI scripting but I was hoping there was a built-in dialplan means for doing this. I have used VMAuthenticate but I would like more flexibility than what this offers What do you need that VMAuthenticate does not offer? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting span orders with Astribank and Sangoma A101
On 12/06/2012, at 12:00 AM, Tzafrir Cohen wrote: On Sun, Jun 10, 2012 at 10:10:29PM +1200, Duncan Turnbull wrote: Hi All Just a quick check on the best way to ensure multiple cards/devices load in the correct order. Asterisk 1.8 with Sangoma A101 had no problems until we introduced an Astribank. root@pabx377:/etc/asterisk# dahdi_hardware -v usb:001/004 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware LABEL=[usb:X1060395] CONNECTOR=@usb-:00:1d.7-3 XBUS-00/XPD-00: FXS (14) Span 2 XBUS-00/XPD-10: FXS (8) Span 3 XBUS-00/XPD-20: FXS (8) Span 4 XBUS-00/XPD-30: FXS (8) Span 5 pci::05:00.0 wanpipe- 1923:0300 Sangoma Technologies Corp. A101 single-port T1/E1 But now they have changed order after reboot to 1-4 and I am guessing that makes the A101 span 5 but I want to make these setting permanent I see mention of /etc/dahdi/xpp_order and dahdi_genconf xpporder but will that help me set the spans within Dahdi so they always appear on the same number The word load is not the correct one here. The Astribank's span will only appear once you run 'dahdi_registration on' . Normally this happens in the dahdi init script, but you can tweak it to happen elsewhere. Another option: if Dahdi 2.6 is an option (no idea if Sangoma supports it): that version has an option to reserve span numbers explicitly for specific hardware devices. With this we can do away with relying on the load order. This is considered experimental (and I wonder how its scriptary interacts with the Sangoma scriptary). We managed to get them booting in the right order but its probably luck at this point, but It seems either the Astribank and the Sangoma play badly together or the Astribank we have is not well. From a dahdi perspective the Astribank looks well, I can see calls to the unit, the lights come on, and xpp_blink works well but we have lost voltage in the unit, the extensions cannot break dial tone or answer a call. So on the electrical side it appears to be failing. http://docs.tzafrir.org.il/dahdi-linux/#_span_assignments , but there are better ways. Let me know what you want to do. We have wired it out using the connector, and were able to plug in an RJ45 in the front but we now can't I don't know if this is somehow related to interaction with the sangoma, the astribank itself failing or otherwise, but it has made my span problem moot for now. We are going back this morning to do more tests and possibly put it on its own asterisk to see if that helps. We will also disconnect it from the wiring connector to see if somehow one of the extensions is doing something unpleasant But any tips would be great. Apologies for changing the topic but I can't find anything out there to give me a clue to an electrical issue. All wisdom appreciated Thanks very much -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which Digium cards to select: AEX with EF or E or P or B?
Dears; I need to order Digium card and not able to know which one is the best quality? Is it that of AEX with the end E or EF or P or B? I saw those card that its slot is small (I think those that end by EF), are they the best card? Really I am caring to have a card that has echo cancelation and the voice volume is high enough (because previously I faced the problem that the voice volume was low and I tried to resolve it without any success). Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between PBX and SBC
That's my question...the sbc provides security over trunking, right? The same can do Asterisk or a Proxy..isn't? Does an SBC can provide any kind of add-value to an Asterisk deployment? El 11/06/2012 20:20, Paul Belanger paul.belan...@polybeacon.com escribió: On 12-06-11 02:06 PM, Danny Dias wrote: Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment What are you expecting the SBC to do? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which Digium cards to select: AEX with EF or E or P or B?
On 06/11/2012 01:48 PM, bilal ghayyad wrote:
I need to order Digium card and not able to know which one is the best quality?
Is it that of AEX with the end E or EF or P or B?
Digium does not produce cards of differing quality levels; we strive to
have all of our cards be produced with the best quality possible.
The various suffixes you refer to indicate whether the board is a plain
board, with modules, with an echo canceller, or some combination of all
of them.
I saw those card that its slot is small (I think those that end by EF), are
they the best card?
Each variant of a particular model is the same size. All AEX410 cards
are the same size, for example, regardless of which modules are
installed or whether an echo canceller is included. Whether a card's
size makes it the 'best' card for your application is up to you to
decide, based on the system you plan to install the card into.
Really I am caring to have a card that has echo cancelation and the voice
volume is high enough (because previously I faced the problem that the voice
volume was low and I tried to resolve it without any success).
It will not matter which Digium card you choose, they will all produce
identical signal levels ('voice volume') when plugged into your
telephony circuit(s).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
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[asterisk-users] Asterisk 1.8.10
Hello, How to change ring tone on interncal call? I'm using Centos 5.8 Asterisk 1.8 exten = 666,1,SIPAddHeader(Alert-Info:http://1.2.3.4/ringtones/ghost.wav) exten = 666,n,Dial(SIP/10) The above would not how to defirenciate from internal call or external call? Thanks, motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.10
Hi As far as I know Alert-Info is as far as vendor specific extension to SIP used by CISCO VOIP-gateways only. Didn't noticed any other vendors to support that. Software clients neither. So such trick is only usable in conjunction with CISCO. Anyway, wait another answer, probably somebody knows more. 2012/6/12 motty.cruz motty.c...@gmail.com: Hello, How to change ring tone on interncal call? I'm using Centos 5.8 Asterisk 1.8 exten = 666,1,SIPAddHeader(Alert-Info:http://1.2.3.4/ringtones/ghost.wav) exten = 666,n,Dial(SIP/10) The above would not how to defirenciate from internal call or external call? Thanks, motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.3.2: Attended transfer goes to incorrect voicemail
On Wed, 6 Jun 2012 16:57:11 -0400 Hai Nguyen h...@jazinga.com wrote: A calls B. B attended-transfers the call to C using (polycom, cisco) phone's transfer button. C does not answer the call. A gets B's voicemail. However, if B blind-transferred the call to C and C did not answer the call, A would get C's voicemail, as expected. With an attended transfer, I would expect B to get C's voicemail, then hang up, resume the call with A and tell them that C is not available. A wouldn't get anyone's voicemail unless it was a blind transfer. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.10
On Mon, Jun 11, 2012 at 6:12 PM, motty.cruz motty.c...@gmail.com wrote: Hello, How to change ring tone on interncal call? I'm using Centos 5.8 Asterisk 1.8 exten = 666,1,SIPAddHeader(Alert-Info:http://1.2.3.4/ringtones/ghost.wav) exten = 666,n,Dial(SIP/10) The above would not how to defirenciate from internal call or external call? Just a thought, but maybe set a variable in your sip.conf for each internal peer, and then check for that variable before you do the SipAddHeader command (using an ExecIf statement). -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
