Re: [asterisk-users] QoS VPN
Access-list 100 permit ip host asterisk server any Class-map match-any voip Match access-group 100 Policy-map voip Class voip Priority 256 Class class-default Fair-queue Interface fastethernet 0 Service-policy output voip Above is what I do to prioritize 256kbit of outbound bandwidth to voip calls, adjust accordingly. You must also use the qos pre-classify in your ipsec tunnel definitions for this to work, but it does work well. I know I'm potentially mapping other traffic than voip, but I'm lazy and don't want to classify the rtp and sip and iax ports, rarely does the box do any other traffic than voip as updates occur in off hours. You'll probably additionally want to match your ipsec keying traffic and give it priority bandwidth, if you're going to push voip through the tunnel you'll find yourself rekeying more often and want to make sure on a saturated link it gets priority so the tunnels don't drop. If you're on DSL, you probably want to research cascading the Qos, have a root policy that throttles all bandwidth to a certain speed, then a child policy that prioritizes that bandwidth, so you don't saturate your outbound circuit(think in terms of P2P protections). This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wanpipe
Newest wanpipe (3.3.16) beta drivers do not compile against dahdi-linux 2.2.0-rc2 which is what you get when you get dahdi-linux-current.tar.gz Anyone have a workaround or patch? Error below Building modules, stage 2. MODPOST CC /usr/src/wanpipe-3.3.16/patches/kdrivers/wanec/wanec.mod.o LD [M] /usr/src/wanpipe-3.3.16/patches/kdrivers/wanec/wanec.ko make[1]: Leaving directory `/usr/src/kernels/2.6.18-128.1.6.el5-x86_64' make -C /lib/modules/2.6.18-128.1.6.el5/build SUBDIRS=/usr/src/wanpipe-3.3.16/kdrvtmp CC=gcc KBUILD_VERBOSE=0 modules make[1]: Entering directory `/usr/src/kernels/2.6.18-128.1.6.el5-x86_64' CC [M] /usr/src/wanpipe-3.3.16/kdrvtmp/sdla_tdmv.o /usr/src/wanpipe-3.3.16/kdrvtmp/sdla_tdmv.c: In function âwp_tdmv_software_initâ: /usr/src/wanpipe-3.3.16/kdrvtmp/sdla_tdmv.c:1097: error: âstruct dahdi_spanâ has no member named âechocanâ make[2]: *** [/usr/src/wanpipe-3.3.16/kdrvtmp/sdla_tdmv.o] Error 1 make[1]: *** [_module_/usr/src/wanpipe-3.3.16/kdrvtmp] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-128.1.6.el5-x86_64' Jeremy Mann This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MySQL queries
I'm running some mysql queries on the standard sql logging of calls, and am interested if anyone has any they'd like to share to get good statistics. I'm interested in # of calls per day, based on DST. Number of Calls per day based on SRC, avg duration of calls, etc.. Thanks. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jm...@txhmg.com This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hacked
Just FYI: IP address 89.248.168.176 has been trying to use the recently release SIP vulnerability in Asterisk to make outbound calls via our box. They are running a bank account callback scam. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jm...@txhmg.com This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Executive Assistant Guidance
Looking for two things: 1. Anyone that has dialplan logic for an executive assistant. My owners want their extensions to ring on her phone, and be very obvious to her which extension is ringing. They also want her to have presense. She's got Polycom IP 650 with sidecar, they have IP 550 phones. Thusfar I've got her registering to 4 extensions. Each extension is labeled with an executive and rings alongside theirs(Dial(SIP/126SIP/191)) just didn't know if there was a better way. I also have presense setup on her Sidecar but it only has one status, is there a way for her to know their line is ringing and not just in use. ? 2. Sort of tied to #1, does anyone have clear dialplan logic and polycom config information about doing custom ringing per extension on the IP 650 ? Thanks. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jm...@txhmg.com This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dundi Issues
I'm getting the following error over and over on the console: pbx_dundi.c:2975 dundi_rexmit: Max retries exceeded to host Any idea how to troubleshoot this? My network latency is roughly 40-50ms between all hosts in my dundi cloud. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: [EMAIL PROTECTED] This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dundi Issues
I don't know if it's related, but when doing a packet sniff with wireshark, I see UDP checksum incorrect messages: 0.058230 source - destination UDP Source port: 4520 Destination port: 4520 [UDP CHECKSUM INCORRECT] The reply packet however does not have this warning: 9.199240 destination - source UDP Source port: 4520 Destination port: 4520 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Wednesday, November 05, 2008 8:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Dundi Issues I'm getting the following error over and over on the console: pbx_dundi.c:2975 dundi_rexmit: Max retries exceeded to host Any idea how to troubleshoot this? My network latency is roughly 40-50ms between all hosts in my dundi cloud. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: [EMAIL PROTECTED] This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dundi Issues
I'm not aware of any offloading done on this particular box, it's an HP ML110 G5 using the onboard NIC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, November 05, 2008 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dundi Issues Jeremy Mann wrote: I don't know if it's related, but when doing a packet sniff with wireshark, I see UDP checksum incorrect messages: 0.058230 source - destination UDP Source port: 4520 Destination port: 4520 [UDP CHECKSUM INCORRECT] Be careful with this error, some network cards that can do IP Offload processing will show up with bad checksums in Wireshark. Check the specs for your NIC, this may be a Red Herring (or it might not! ). regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] users.conf and hints
Is there a way to override sip peers defined in users.conf with respect to their context and hints? Every extension I have defined in users.conf always gets an @default for the hint priority. Below are asterisk outputs and users.conf entries. In peer 1203 I've set a subscribecontext, which is completely ignored. Thanks for any help. nurscarepbx*CLI core show version Asterisk 1.4.22 built by root @ nurscarepbx on a x86_64 running Linux on 2008-10-16 12:37:36 UTC Pbx*CLI show hints ... [EMAIL PROTECTED] : SIP/1203 State:Idle Watchers 0 [EMAIL PROTECTED] : SIP/1202 State:Idle Watchers 0 ... Users.conf [1202] fullname = 1202 secret = 1202 hasvoicemail = yes mailbox = [EMAIL PROTECTED] vmsecret = 1234 hassip = yes hasmanager = no callwaiting = no context = from-nortel call-limit = 4 dynamic = yes qualify = yes host = dynamic [1203] fullname = 1203 secret = 1203 hasvoicemail = yes mailbox = [EMAIL PROTECTED] vmsecret = 1234 hassip = yes hasmanager = no callwaiting = no context = from-nortel subscribecontext = internal call-limit = 4 dynamic = yes qualify = yes host = dynamic Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: [EMAIL PROTECTED] This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma Question
Any advise on troubleshooting this: Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: OOF alarm is OFF Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: RED alarm is OFF Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RAI alarm is OFF Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RED alarm is OFF It happens nightly, and I have to reset asterisk to clear it. Zap/Dahdi channels wont' work until I do. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: [EMAIL PROTECTED] This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Headset Recommendation
Does anyone have a recommendation for a headset that plugs into the Mic/Line-out port on a PC? Ideally something like the Plantronics SupraPlus. I'd prefer Monaural instead of stereo, and cheap in price but not in quality. Thanks for any suggestions... Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: [EMAIL PROTECTED] This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] users.conf and sip call-limit
Does the call-limit directive work on those SIP items defined in users.conf as it relates to presence and queues? Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: [EMAIL PROTECTED] This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
Tried using GROUP()?
When a call comes in or goes out:
Exten = XXX,1,Set(GROUP(bdwi_out_1)=outgoing/incoming);
Exten = XXX,n,GotoIf($[${GROUP_COUNT(outgoing/[EMAIL PROTECTED])}] 1?fail)
Exten = XXX,n,Dial(...)
Exten = XXX(fail),1,Congestion();
Exten = XXX(fail),n,Hangup();
Obviously choose outgoing or incoming, if you want to track both you can just
use $MATH() to add them together.
Or some other math logic to check the result.
On incoming Set(DIALSTATUS=CHANUNAVAIL) and it'll ring busy to bandwidth(or out
of service, you can tweak this).
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Knudsen
Sent: Monday, October 20, 2008 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent
one-way audio
Any updates? It still seems to happen, though not as often as it used to. We're
using Polycom 320 phones, if that makes a difference, though we did do it with
X-Lite as well.
On Sat, Oct 11, 2008 at 3:03 PM, Kurt Knudsen [EMAIL PROTECTED]mailto:[EMAIL
PROTECTED] wrote:
Thanks, Steve,
That's what I am unsure of. I don't know how to limit 1 call per trunk. If
that's an easy thing to setup, I'd love to see it.
On Fri, Oct 10, 2008 at 10:20 PM, Steve Totaro [EMAIL PROTECTED]mailto:[EMAIL
PROTECTED] wrote:
Oh, I thought you had logic to count the calls on the trunk. You should limit
each trunk to one call. This is the primary reason besides the email that
basically said that customer support structure has been changed and anything
beyond the Demarc would not be supported, I the Demarc is simply their boxen,
so unless it is on their side, you will not get any helpful support from
Bandwidth, plus they CCed over 500 people by address instead of setting up a
group. http://www.bandwidth.com/content/support/?page=standardSupport
I am with Junction and while a trunk is not unlimited as far as price for
usage, the amount of trunks is unlimited (or at least as unlimited as it can be
since nothing is really unlimited). They asked that I try not to go over one
call per second for any real duration, and that I not hammer one LATA do to
limited interconnects.
The other thing was Junctions was very easy to sign up with, great support, and
configuration was a breeze.
As for Bandwidth, I think they are solid but due to recent changes and the fact
that you must pay per channel, as well as the setup process, I decided they
were not for me.
I will take a second look at your sip.conf and extensions.conf later to see if
something jumps out at me. I suspect since you are setting up two separate
trunks with Bandwidth, you need to limit each trunk to one call, rather than
two.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 9:47 PM, Kurt Knudsen [EMAIL PROTECTED]mailto:[EMAIL
PROTECTED] wrote:
externip messes up DTMF detection, and by messes up I mean it doesn't detect it
at all. Setting nat=yes or nat=no didn't make a difference either.
When the trunks are in use, the calls are fine, no dropped audio. It only
happens when a 3rd call is made and there's no trunk available.
Thanks :)
On Fri, Oct 10, 2008 at 7:09 PM, Steve Totaro [EMAIL PROTECTED]mailto:[EMAIL
PROTECTED] wrote:
You need to configure your box for nat settings, externip and other settings in
sip.conf and set nat=yes instead of nat=no.
One way audio is almost always a NAT issue and those are two glaring things
that would cause problems.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen [EMAIL PROTECTED]mailto:[EMAIL
PROTECTED] wrote:
Hi Steve,
It's behind a NAT/Firewall but SIP translation is enabled and removing it from
behind the firewall did nothing, it still dropped calls. The calls connect and
everything works, but it dies when all trunks are in use and someone else tries
to call out. It seems like even though both channels are in use, it tries to
connect to the 2nd trunk and thus kills the audio. Nothing strange came up in
Wireshark or the firewall logs.
Thanks.
On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro [EMAIL PROTECTED]mailto:[EMAIL
PROTECTED] wrote:
On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen [EMAIL PROTECTED]mailto:[EMAIL
PROTECTED] wrote:
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them). This happens 100% of the time and Bandwidth.com
doesn't offer any support. I don't see any setting that tells Asterisk that
there are 2 channels available from Bandwidth.com's IP. I'm currently using, or
attempting to use, groups to solve this problem, but sometimes it works,
sometimes it doesn't. It breaks when a call goes out on a Queue, because it
seems to add each phone to the group, which breaks my GotoIf() statement.
Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI
Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
My experience with GotoIf, what follows the ? has to be part of the extension
itself.
In your example:
Exten = _1NXXNXX(100) would be the intended target.
Maybe that's just 1.4 specific, I'll admit I haven't read this entire thread.
Also, use specific groups:
Set(GROUP(SIP)=SIPGROUP)
Set(GROUP(SIP_PHONE)=SIPGROUP)
Those are two distinct ways to track them, instead of a general GROUP()
statement. Since a channel can only be a member of one GROUP(), but multiple
GROUP(XXX) it makes it easier to track items when they belong to multiple
things(and logically reads better for future support of the dialplan).
I can say that we're successfully limiting calls on two-way sip trunks from
bandwidth, both incoming and outgoing.
Probably if 4+ lit up at once I'd have a problem, but we're not that high
volume.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Knudsen
Sent: Monday, October 20, 2008 11:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent
one-way audio
I tried using GROUP(), here's a snippet from the first post.
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix
it added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as backup.
;This is where it breaks. I tried to make it so there can't be more
than 2 calls on SIP channels at once.
;Since it counts the phone as a channel, and adds it to the group, I
had to use 4.
[internalphones]
exten = _1NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _1NXXNXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} = 4]?100)
;If the group has 2 or more calls, do not dial.
exten = _1NXXNXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})
exten =
_1NXXNXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)
exten = _1NXXNXX,100,Playback(all-circuits-busy-now)
exten = _1NXXNXX,101,congestion()
exten = _1NXXNXX,102,busy()
;This is where incoming calls go to if I'm awake.
[DID_trunk_2_timeinterval_Awake]
exten = _NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _NXXNXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})
exten = _NXXNXX,n,Set(CALLERID(num)=1${CALLERID(num)})
exten = _NXXNXX,n,Goto(voicemenu-custom-1|s|1)
I'll try playing around with incoming/outgoing and see if that makes a
difference. I don't know why it counts the phone as a channel, though.
On Mon, Oct 20, 2008 at 12:14 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
Tried using GROUP()?
When a call comes in or goes out:
Exten = XXX,1,Set(GROUP(bdwi_out_1)=outgoing/incoming);
Exten = XXX,n,GotoIf($[${GROUP_COUNT(outgoing/[EMAIL PROTECTED])}] 1?fail)
Exten = XXX,n,Dial(...)
Exten = XXX(fail),1,Congestion();
Exten = XXX(fail),n,Hangup();
Obviously choose outgoing or incoming, if you want to track both you can just
use $MATH() to add them together.
Or some other math logic to check the result.
On incoming Set(DIALSTATUS=CHANUNAVAIL) and it'll ring busy to bandwidth(or
out of service, you can tweak this).
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Knudsen
Sent: Monday, October 20, 2008 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent
one-way audio
Any updates? It still seems to happen, though not as often as it used to.
We're using Polycom 320 phones, if that makes a difference, though we did do
it with X-Lite as well.
On Sat, Oct 11, 2008 at 3:03 PM, Kurt Knudsen [EMAIL PROTECTED] wrote:
Thanks, Steve,
That's what I am unsure of. I don't know how to limit 1 call per trunk. If
that's an easy thing to setup, I'd love to see it.
On Fri, Oct 10, 2008 at 10:20 PM, Steve Totaro [EMAIL PROTECTED] wrote:
Oh, I thought you had logic to count the calls on the trunk. You should
limit each trunk to one call. This is the primary reason besides the email
that basically said that customer support structure has been changed and
anything beyond the Demarc would not be supported, I the Demarc is simply
their boxen, so unless it is on their side, you will not get any helpful
support from Bandwidth, plus they CCed over 500 people by address instead of
setting up a group.
http://www.bandwidth.com/content/support/?page=standardSupport
I am with Junction and while a trunk is not unlimited as far as price for
usage, the amount of trunks is unlimited (or at least as unlimited as it can
be since nothing is really unlimited). They asked that I try not to go over
one call per second for any real duration
Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
I have a macro to dial out, similar to yours in that it fails over to Zap/Dahdi
trunks in the event our bandwidth stuff is overloaded.
I run this in a macro, and only set and check groups within that macro. I'm
confused why yours would attach to phones in any way, unless you mean phone
to phone calls, in that case don't set the group?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Knudsen
Sent: Monday, October 20, 2008 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent
one-way audio
The GotoIf works, because it does failover sometimes, just not all the
time, I followed instructions from here:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf
And it seems to work in other areas that I use it in a similar way. I
only have the Set(GROUP()) when we are making outgoing calls on the
SIP trunk or when there's an incoming call on the SIP trunk. Anything
on Dahdi doesn't get included. I don't know how to tell my phones and
channels apart, I'm not trying to add the phones to the group, just
the channels. Can you paste some of your extensions.conf since you
also use Bandwidth.com?
Thanks.
On Mon, Oct 20, 2008 at 8:30 PM, [EMAIL PROTECTED] wrote:
-- Kurt Knudsen wrote :
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband because we
were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;...irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix it
added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as backup.
;This is where it breaks. I tried to make it so there can't be more than 2
calls on SIP channels at once.
;Since it counts the phone as a channel, and adds it to the group, I had to
use 4.
[internalphones]
exten = _1NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _1NXXNXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} = 4]?100) ;If the
group has 2 or more calls, do not dial.
exten = _1NXXNXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})
exten =
_1NXXNXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)
exten = _1NXXNXX,100,Playback(all-circuits-busy-now)
exten = _1NXXNXX,101,congestion()
exten = _1NXXNXX,102,busy()
;This is where incoming calls go to if I'm awake.
[DID_trunk_2_timeinterval_Awake]
exten = _NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _NXXNXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})
exten = _NXXNXX,n,Set(CALLERID(num)=1${CALLERID(num)})
exten = _NXXNXX,n,Goto(voicemenu-custom-1|s|1)
Thanks.
--
This message was sent on behalf of [EMAIL PROTECTED] at openSubscriber.com
http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/10416933.html
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[asterisk-users] IP 650 Sidecar
Is the IP 650 sidecar compatible with asterisk? If I pair it with the IP 650 phone, can I have more than 6 lines registered w/ the server? Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: [EMAIL PROTECTED] This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parked Calls
Can anyone explain parked calls?
I've run so many tests over the last few hours I'm totally confused. Half the
time the call times out and returns back to the user that dialed it, through
the same context it was originated from.
The other half it returns to the park-dial context with a dynamically added
context.
I have park-dial defined as:
context park-dial {
s = {
jump [EMAIL PROTECTED];
};
_. = {
jump [EMAIL PROTECTED];
};
t = {
jump [EMAIL PROTECTED];
};
};
When I dial from extension 155 to extension 698 on this system, park-dial
contains a dynamic SIP/155 extension defined in addition to my three above. It
however never matches the _. extension and instead returns back to the original
caller.
Is there a reliable way to override the position it returns to when timing out
using the default parking setup in features.conf, or would it be easier to do
it with a custom parking extension setup?
If so, anyone have examples of custom parking?
Thanks.
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it contains information that is confidential and privileged. This information
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addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
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Re: [asterisk-users] Parked Calls
Forgot to mention, I'm running asterisk 1.4.21.2
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann
Sent: Wednesday, September 17, 2008 2:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Parked Calls
Can anyone explain parked calls?
I've run so many tests over the last few hours I'm totally confused. Half the
time the call times out and returns back to the user that dialed it, through
the same context it was originated from.
The other half it returns to the park-dial context with a dynamically added
context.
I have park-dial defined as:
context park-dial {
s = {
jump [EMAIL PROTECTED];
};
_. = {
jump [EMAIL PROTECTED];
};
t = {
jump [EMAIL PROTECTED];
};
};
When I dial from extension 155 to extension 698 on this system, park-dial
contains a dynamic SIP/155 extension defined in addition to my three above. It
however never matches the _. extension and instead returns back to the original
caller.
Is there a reliable way to override the position it returns to when timing out
using the default parking setup in features.conf, or would it be easier to do
it with a custom parking extension setup?
If so, anyone have examples of custom parking?
Thanks.
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
its subsidiaries, and affiliates hereby claim all applicable privileges related
to this information.
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is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
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you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
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[asterisk-users] Parked Calls
Using the default features.conf setup, if I include parkedcalls in my dialplan,
and a call gets parked, and times out, where does the call go?
Does it go to a timeout extension in parked calls, or does it go to a timeout
extension in the original context?
(Using an AEL based dialplan similar to below).
--
context internal {
...
...
t {
jump [EMAIL PROTECTED];
};
includes {
parkedcalls;
};
};
Jeremy Mann
Director of IT
Texas Health Management Group
Direct Line: 817-310-4956
Main Line: 817-310-4999
Fax: 817-310-4990
Email: [EMAIL PROTECTED]
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
its subsidiaries, and affiliates hereby claim all applicable privileges related
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Re: [asterisk-users] Parked Calls
Which would imply you have parked calls, upon timeout, going to a different context. Where did you define that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, September 16, 2008 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Parked Calls Jeremy Mann wrote: Using the default features.conf setup, if I include parkedcalls in my dialplan, and a call gets parked, and times out, where does the call go? I can't tell you about AEL, but I have the following: [park-dial] ; ; Don't drop unanswered parked ; calls, send them to the operator ; exten = t,1,Goto(office-hours,s,6) Doug This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parked Calls
I just tested without any t extension, and the parked call times out and rolls back to the user that parked it. I need a way to override this to roll back to my operator. Any asterisk gurus that can validate I just need a [park-dial] context, or a way to override the timeout to point to another extension/context? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, September 16, 2008 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Parked Calls Jeremy Mann wrote: Which would imply you have parked calls, upon timeout, going to a different context. Where did you define that? I didn't, I believe it's hard coded. It was being displayed on the console way back in v1.2 about calls being sent to park-dial and no timeout being defined, so I added the [park-dial] context and put a timeout extension. Worked like a charm. 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX over T1 Question
context from-pri {
_8505 = {
Wait(1);
Answer();
SetTransferCapability(3K1AUDIO);
Set(GROUP(ZAP)=incoming);
Set(CDR(accountcode)=fax);
Set(CDR(userfield)=bedford);
Dial(Zap/25/${EXTEN});
Hangup();
The above AEL logic is my DID for faxing. The card wasn't detecting the
3K1AUDIO (digium TE205P rev 02) so I set it manually. Zap/25 is on the 2nd
port of the T1 connected to a channel bank.
It also appears to make it disable EC no matter what channel of the PRI the fax
comes in on.
Maybe it does nothing, but it immediately fixed faxing whereas before we were
having intermittent issues.
I also have that set on in-outbound modem calls from specific channels on the
channel bank, they are getting as close to 56k as I've ever seen.
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it contains information that is confidential and privileged. This information
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addressed. If you are the intended recipient, further disclosures are
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any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
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[asterisk-users] PRI Splitter
Does anyone know of a pri splitter device? Something that would take an incoming PRI, and based on DID send that out one of other multiple PRI ports? I'm needing to take a single PRI from the telco, and send it to two separate phone systems(one asterisk) based on DID. I know I could probably achieve the same thing with a 3 port PRI card in a server, but I'd like something braindead easy to configure from both a hardware and software perspective. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
We've done the asterisk passthrough route, but if the asterisk box is down for whatever reason both systems are down. Splitter wasn't the right word, but yes I see your point, I'll look into the Adtran. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Wednesday, August 27, 2008 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI Splitter Jeremy Mann wrote: I know I could probably achieve the same thing with a 3 port PRI card in a server, but I'd like something braindead easy to configure from both a hardware and software perspective. Anything you use is going to (essentially) be a 3-port ISDN PRI capable switch, because that is the only way to accomplish what you need. There really isn't any way to 'split' a PRI, unlike a T1 using CAS signaling which can be 'split' using a drop-and-insert multiplexer. If you don't want to use a small PC with a 3-port T1 card in it, you can use something like an Adtran Atlas to do the job. Alternatively, just use a 2-port T1 card in the Asterisk server, and run the PRI *through* the Asterisk server on the way to the other PBX. That's the most common way to do what you want to do. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Channel Oddity
Yes, it's an _X. match for local/ld It actually ended up being oddity with Centos 5.2, I had to upgrade Zaptel to the newest version and it resolved it, apparently it wasn't passing all the digits to the line. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dana Harding Sent: Wednesday, July 16, 2008 4:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Channel Oddity - Original Message - Can anyone help me start to diagnose why a Sangoma A200 wouldn't dial out LD? Local calls are fine, incoming is fine, just no LD. Bell tech has been on site and plugged into lines with his test set and was able to dial LD just fine, so it's not a LEC issue. No errors in asterisk console, using zaptel 1.4.11 and sangoma drivers 3.2.6, asterisk 1.4.18 What does your dialplan (extensions.conf) look like for outgoing calls - is there a matching extension for LD calls (exten = _1NXXNXX,)? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap Channel Oddity
Can anyone help me start to diagnose why a Sangoma A200 wouldn't dial out LD? Local calls are fine, incoming is fine, just no LD. Bell tech has been on site and plugged into lines with his test set and was able to dial LD just fine, so it's not a LEC issue. No errors in asterisk console, using zaptel 1.4.11 and sangoma drivers 3.2.6, asterisk 1.4.18 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap Bridged Channels
I have a Sangoma A200DX, and am trying to bridge an FXO channel with FXS for modem connectivity. I have Zap/8 as a Fax Machine Zap/5 is my outside line. When a call rings in on Zap/5 it immediately calls Zap/8 and bridges the channels. I see it doing a native bridge on the two. I have echo cancel off on native bridge, but I can never get fax connectivity, it just tries to negotiate forever then eventually hangs up. Anything special to getting this to work? Below is an example of CLI output when the Fax Machine tries to call out, it does the same thing, never get the two machines to complete the call and send the fax. I've also included the CLI output of channel 5's properties, it does show the EC as off. I noticed it says Fax Handled: no, is there something I need to enable in Zapata.conf or zaptel.conf? Would txgain/rxgain be the issue? CLI Output -- Starting simple switch on 'Zap/8-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/8-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/8-1, Zap/5) in new stack -- Called 5 -- Zap/5-1 is ringing -- Zap/5-1 is ringing -- Zap/5-1 answered Zap/8-1 -- Native bridging Zap/8-1 and Zap/5-1 localhost*CLI zap show channel 5 Channel: 5CLI File Descriptor: 27 Span: 2 Extension: Dialing: no Context: from-internal-fax Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: FXO Kewlstart Radio: 0 Owner: Zap/5-1 Real: Zap/5-1 Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF Master Channel: 8 Actual Confinfo: Num/8, Mode/0x0009 Actual Confmute: No Hookstate (FXS only): Onhook Zapata.conf - [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no relaxdtmf=yes rxgain=2.0 txgain=2.0 group=1 callgroup=1 pickupgroup=1 immediate=no context=from-internal-fax group=1 signalling = fxo_ks channel = 5 context=from-zaptel-fax group=3 signalling = fxs_ks channel = 8 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Bridged Channels
I set it up in general because my voice lines(ports 1-4) had very low volume, and callers complained about outgoing as well, upping both to two seemed to resolve them. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, July 09, 2008 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Bridged Channels On Wed, Jul 9, 2008 at 3:28 PM, Jeremy Mann [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] wrote: I have a Sangoma A200DX, and am trying to bridge an FXO channel with FXS for modem connectivity. I have Zap/8 as a Fax Machine Zap/5 is my outside line. When a call rings in on Zap/5 it immediately calls Zap/8 and bridges the channels. I see it doing a native bridge on the two. I have echo cancel off on native bridge, but I can never get fax connectivity, it just tries to negotiate forever then eventually hangs up. Anything special to getting this to work? Below is an example of CLI output when the Fax Machine tries to call out, it does the same thing, never get the two machines to complete the call and send the fax. I've also included the CLI output of channel 5's properties, it does show the EC as off. I noticed it says Fax Handled: no, is there something I need to enable in Zapata.conf or zaptel.conf? Would txgain/rxgain be the issue? Gain certainly could be an issue. Did you set them to 2 for a reason? If not try 0 gain. I suspect if you set the communication speed on the fax to a slow speed it will work. 9600 and then bump it up. Thanks, Steve T This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL Help
I need help translating extensions.conf to AEL:
[default]
exten = _X.,1,Set(DID=${EXTEN:6})
exten = _X.,n,Goto(continue,1)
exten = _1X.,1,Set(DID=${EXTEN:7})
exten = _1X.,n,Goto(continue,1)
exten = continue,1,Noop(${DID})
exten = continue,n,Set(GROUP(IAX)=incoming)
exten = continue,n,GotoIf($[${MATH(${GROUP_COUNT([EMAIL
PROTECTED])}+${GROUP_COUNT([EMAIL PROTECTED])},i)}10]?fail)
exten = continue,n,Goto(from-pri,${DID},1)
exten = continue,n(fail),Set(DIALSTATUS=CHANUNAVAIL)
I need the above to goto AEL, here's what I have so far:
context default {
_X. = {
Set(DID=${EXTEN:6});
Goto(continue,1);
};
_1X. = {
Set(DID=${EXTEN:7});
Goto(continue,1);
};
continue:
Noop(${DID});
Set(GROUP(IAX)=incoming);
GotoIf($[${MATH(${GROUP_COUNT([EMAIL
PROTECTED])}+${GROUP_COUNT([EMAIL PROTECTED])},i)}10]?fail);
Goto(from-pri,${DID},1);
fail:
Set(DIALSTATUS=CHANUNAVAIL);
};
};
My issue is I don't know what to do with the fail and continue goto statements.
Thanks.
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any disclosure, copying, printing, or use of this information is strictly
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[asterisk-users] *72 Telco Call Forwarding
Is there a way to force asterisk to ignore the first ring of a call without using Wait() ? When I active *72 call forward on my analog lines from the telco, they always send a single ring and then do the forwarding. Asterisk picks up essentially a dead line and rings the phones which gets really annoying. Thanks. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi and SIP
I think I'm going to go about this a different way, if it works I'll post my
solution.
Essentially I'm going to limit the calls by grouping(didn't know you could use
categories until I did the research) and math. Limiting our corporate office
to 10 IAX calls, both incoming and outgoing together, and denying the call if
it's above that(sending chanunavail or something similar).
I'll then run all dials through a macro, looking up dundi routes. If it fails
I'll fall back to zap.
Thanks for the help though.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, April 23, 2008 5:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Take a look at this setup, it does not use passwords on the sip peers
or the mappings in Dundi. As long as you inside your network this
maybe the way to go.
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords
You could also look at the incominglimit and outgoinglimit on IAX peers
On Wed, Apr 23, 2008 at 4:51 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm fairly sure SIP will never work unless I hard-code peers everywhere,
which isn't going to happen. The only reason I want to use it is for the
call-limit option.
Looking at sip channels there is no option to pass the extension after the
IP, it's always [EMAIL PROTECTED], or [EMAIL PROTECTED], not [EMAIL
PROTECTED]/extension or [EMAIL PROTECTED]/extension
Looks like IAX and ZAP are the only two channel types that do a /extension
type setup.
Extensions.conf:
[macro-dundi-lookup]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-local
include = dundi-priv-lookup
[dundi-priv-local]
include = internal
[dundi-priv-lookup]
switch = DUNDi/priv
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, April 23, 2008 4:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
It is not the dip peer that is failing but the dial plan:
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
host: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
== Everyone is busy/congested at this time (1:0/0/1)
What is in the context macro-dundi-lookup?
On Wed, Apr 23, 2008 at 12:47 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
Nope..
asterisk*CLI dundi lookup [EMAIL PROTECTED]
1. 0 SIP/priv:[EMAIL PROTECTED]/400 (EXISTS)
from 00:1e:0b:dd:e9:99, expires in 5 s
DUNDi lookup completed in 104 ms
-- Executing [EMAIL PROTECTED]:1] Set(SIP/156-08274b60,
CDR(accountcode)=wth) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/156-08274b60,
CALLERID(all)=Corporate 100) in new stack
-- Executing [EMAIL PROTECTED]:3] Macro(SIP/156-08274b60,
dundi-lookup|400) in new stack
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/156-08274b60, 400|1) in
new stack
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
host: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:4] Hangup(SIP/156-08274b60, ) in
new stack
== Spawn extension (from-sip, 400, 4) exited non-zero on
'SIP/156-08274b60'
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Tuesday, April 22, 2008 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Try this,
[priv]
dbsecret=dundi/secret
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal
type=friend
priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I
can't specify
[asterisk-users] Macro/Goto Help
I have a macro that checks to see if a dundi route exists, if it does it
attempts to dial it. The remote end can set the chan as unavailable, or busy.
If it does the call immediately hangs up instead of returning to the macro for
more processing. Is there a way to force it to return?
Logic from extensions.conf is below, first is from the system making the call,
the second is from the system receiving the call:
(CALLING SYSTEM)
The DUNDi system makes calls via IAX using a peer named priv
[local-dundi]
exten = _817NXX,1,Macro(dundi-lookup,${EXTEN})
exten = _817NXX,n,Macro(trunkdial,Zap/G0/w${EXTEN})
exten = _NXXNXX,1,Macro(trunkdial,Zap/G0/w${EXTEN})
[macro-dundi-lookup]
exten = s,1,Goto(${ARG1},1)
exten = s,n,MacroExit
include = dundi-priv-local
include = dundi-priv-lookup
include = dundi-e164-lookup
[dundi-priv-local]
exten = _4XX,1,Noop
[dundi-priv-lookup]
switch = DUNDi/priv
[dundi-e164-lookup]
switch = DUNDi/e164
(CALLED SYSTEM)
The IAX peer priv is dropped into the following context in the dialplan
[dundi-e164]
exten = _817.,1,Set(DID=${EXTEN:6})
exten = _817.,n,Noop(${DID})
exten = _817.,n,Set(GROUP(IAX)=incoming)
exten = _817.,n,GotoIf($[${MATH(${GROUP_COUNT([EMAIL
PROTECTED])}+${GROUP_COUNT([EMAIL PROTECTED])},i)}10]?fail)
exten = _817.,n,Goto(from-pri,${DID},1)
exten = _817.,n(fail),Set(DIALSTATUS=CHANUNAVAIL)
If the total for all IAX calls is above 10, I want the call to fail so it'll
fall back and use ZAP instead of IAX. Instead the call just hangs up at the
CALLING system.
The from-pri logic has been excluded since it has no bearing on the question at
hand.
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
its subsidiaries, and affiliates hereby claim all applicable privileges related
to this information.
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Re: [asterisk-users] Macro/Goto Help
Nevermind, helps when you reload the diaplan at BOTH ends :)
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann
Sent: Thursday, April 24, 2008 9:48 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Macro/Goto Help
I have a macro that checks to see if a dundi route exists, if it does it
attempts to dial it. The remote end can set the chan as unavailable, or busy.
If it does the call immediately hangs up instead of returning to the macro for
more processing. Is there a way to force it to return?
Logic from extensions.conf is below, first is from the system making the call,
the second is from the system receiving the call:
(CALLING SYSTEM)
The DUNDi system makes calls via IAX using a peer named priv
[local-dundi]
exten = _817NXX,1,Macro(dundi-lookup,${EXTEN})
exten = _817NXX,n,Macro(trunkdial,Zap/G0/w${EXTEN})
exten = _NXXNXX,1,Macro(trunkdial,Zap/G0/w${EXTEN})
[macro-dundi-lookup]
exten = s,1,Goto(${ARG1},1)
exten = s,n,MacroExit
include = dundi-priv-local
include = dundi-priv-lookup
include = dundi-e164-lookup
[dundi-priv-local]
exten = _4XX,1,Noop
[dundi-priv-lookup]
switch = DUNDi/priv
[dundi-e164-lookup]
switch = DUNDi/e164
(CALLED SYSTEM)
The IAX peer priv is dropped into the following context in the dialplan
[dundi-e164]
exten = _817.,1,Set(DID=${EXTEN:6})
exten = _817.,n,Noop(${DID})
exten = _817.,n,Set(GROUP(IAX)=incoming)
exten = _817.,n,GotoIf($[${MATH(${GROUP_COUNT([EMAIL
PROTECTED])}+${GROUP_COUNT([EMAIL PROTECTED])},i)}10]?fail)
exten = _817.,n,Goto(from-pri,${DID},1)
exten = _817.,n(fail),Set(DIALSTATUS=CHANUNAVAIL)
If the total for all IAX calls is above 10, I want the call to fail so it'll
fall back and use ZAP instead of IAX. Instead the call just hangs up at the
CALLING system.
The from-pri logic has been excluded since it has no bearing on the question at
hand.
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
its subsidiaries, and affiliates hereby claim all applicable privileges related
to this information.
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
its subsidiaries, and affiliates hereby claim all applicable privileges related
to this information.
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Re: [asterisk-users] DUNDi and SIP
Nope..
asterisk*CLI dundi lookup [EMAIL PROTECTED]
1. 0 SIP/priv:[EMAIL PROTECTED]/400 (EXISTS)
from 00:1e:0b:dd:e9:99, expires in 5 s
DUNDi lookup completed in 104 ms
-- Executing [EMAIL PROTECTED]:1] Set(SIP/156-08274b60,
CDR(accountcode)=wth) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/156-08274b60,
CALLERID(all)=Corporate 100) in new stack
-- Executing [EMAIL PROTECTED]:3] Macro(SIP/156-08274b60,
dundi-lookup|400) in new stack
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/156-08274b60, 400|1) in new
stack
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such host:
192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:4] Hangup(SIP/156-08274b60, ) in new
stack
== Spawn extension (from-sip, 400, 4) exited non-zero on 'SIP/156-08274b60'
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Try this,
[priv]
dbsecret=dundi/secret
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal
type=friend
priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I can't
specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box
is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL
PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
This e-mail, facsimile, or letter and any files or attachments
Re: [asterisk-users] DUNDi and SIP
I'm fairly sure SIP will never work unless I hard-code peers everywhere, which
isn't going to happen. The only reason I want to use it is for the call-limit
option.
Looking at sip channels there is no option to pass the extension after the IP,
it's always [EMAIL PROTECTED], or [EMAIL PROTECTED], not [EMAIL
PROTECTED]/extension or [EMAIL PROTECTED]/extension
Looks like IAX and ZAP are the only two channel types that do a /extension type
setup.
Extensions.conf:
[macro-dundi-lookup]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-local
include = dundi-priv-lookup
[dundi-priv-local]
include = internal
[dundi-priv-lookup]
switch = DUNDi/priv
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, April 23, 2008 4:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
It is not the dip peer that is failing but the dial plan:
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
host: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
== Everyone is busy/congested at this time (1:0/0/1)
What is in the context macro-dundi-lookup?
On Wed, Apr 23, 2008 at 12:47 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
Nope..
asterisk*CLI dundi lookup [EMAIL PROTECTED]
1. 0 SIP/priv:[EMAIL PROTECTED]/400 (EXISTS)
from 00:1e:0b:dd:e9:99, expires in 5 s
DUNDi lookup completed in 104 ms
-- Executing [EMAIL PROTECTED]:1] Set(SIP/156-08274b60,
CDR(accountcode)=wth) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/156-08274b60,
CALLERID(all)=Corporate 100) in new stack
-- Executing [EMAIL PROTECTED]:3] Macro(SIP/156-08274b60,
dundi-lookup|400) in new stack
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/156-08274b60, 400|1) in
new stack
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such host:
192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:4] Hangup(SIP/156-08274b60, ) in new
stack
== Spawn extension (from-sip, 400, 4) exited non-zero on 'SIP/156-08274b60'
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Try this,
[priv]
dbsecret=dundi/secret
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal
type=friend
priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I
can't specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL
PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure
Re: [asterisk-users] DUNDi and SIP
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I can't
specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and
entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of federal
or state law and regulations. If you have received this information in
error, please notify Texas Health Management Group immediately at
1-817-310-4999. Texas Health Management Group, its subsidiaries, and
affiliates hereby claim all applicable privileges related to this
information.
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--
*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
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with it contains information that is confidential and privileged. This
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Re: [asterisk-users] Question on groups
Try GROUP()=internal-...
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Friday, April 18, 2008 11:30 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Question on groups
I believe I am close to fixing my problems with my 1.2 to 1.4.19 upgrade. I
have one question: to limit my customers to the number of channels they have
paid for, I use the GROUP feature. I also regularly check in the CLI what`s
going on using group show channels.
Basically, my system is designed so that an external call is set to group
$ACCOUNTCODE and an internal call set to group internal-$ACCOUNTCODE.
i.e.: 5551234567 for an external call and internal-5551234567 for an internal
call. Internal calls are not limited, but I still keep track of them.
When I used group show channels in the 1.2 CLI, I`d get a healthy mix of
5551234567 and internal-5551234567. Ever since moving to 1.4.19, I only see
the external calls when using group show channels. Nothing about the internal
calls.
The relevant line is in my extensions.conf is :
exten = _X!.,n,Set(GROUP=internal-${CDR(accountcode)})
And the CLI shows :
-- Executing [EMAIL PROTECTED]:2] Set(SIP/00041234432-1-b7d4b908,
GROUP=internal-5149070849) in new stack
Which seems right. But it never shows up in the CLI when checking for group
channels.
Any clue? Did something major change between 1.2 and 1.4?
Mike
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
its subsidiaries, and affiliates hereby claim all applicable privileges related
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Re: [asterisk-users] DUNDi and SIP
I have it working via IAX, when I try changing everything to SIP I can't
specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and
entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of federal
or state law and regulations. If you have received this information in
error, please notify Texas Health Management Group immediately at
1-817-310-4999. Texas Health Management Group, its subsidiaries, and
affiliates hereby claim all applicable privileges related to this
information.
___
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--
*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
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prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
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[asterisk-users] users.conf and voicemail
Is there a way to specify per user attachment options for voicemail, from within users.conf? I know I can enable or disable it globally in voicemail.conf, but I have certain users that like the attachment feature, and others that don't. Also, can you enable/disable per user the deletion if it's attached? Thanks. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi and SIP
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]mailto:SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and pass
the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer a.b.c.d/${NUMBER}
cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip peer
per box(I have 25 connected in my dundi cloud), nor would I like to enable
anonymous SIP calls, as I have the ports open to the world for inbound sip from
bandwidth.com
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
its subsidiaries, and affiliates hereby claim all applicable privileges related
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Re: [asterisk-users] Zap Codec
But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin Sent: Monday, April 14, 2008 9:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec This is SIP channel you need to limit. Forcing ulaw only, the Polycom will fall back to ulaw. Per peer, in your sip.conf: disallow=all allow=ulaw From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Monday, April 14, 2008 14:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zap Codec Is there a way to force Zap channels to only use ulaw, and not even attempt g729 negotiation? My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I'm not licensed for the codec on the asterisk box. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Codec
So in other words, if I have G729 enabled on the phones, I must get G729 licenses to use Zap channels. Otherwise I have to use ULAW for everything? I fail to understand why it'd be difficult to do codec negotiation on SIP-ZAP calls, Zap sends that it only supports ulaw, if the phone doesn't then the call is cancelled or forwarded to logic to translate. I realize G729 is fairly cheap, but it's useless server overhead when the phone supports the codec it needs natively. Is there any dialplan logic that can coerce the transaction to be ulaw only? Setting something in the SIP header perhaps? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 8:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec The PSTN only allows ulaw or alaw (depending on your location). You CANNOT send calls in any other codec over a PSTN line. Generally, if you want to use G729 then you must buy a G729 license (with a few exceptions). Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Darryl Dunkin *Sent:* Monday, April 14, 2008 9:01 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Zap Codec This is SIP channel you need to limit. Forcing ulaw only, the Polycom will fall back to ulaw. Per peer, in your sip.conf: disallow=all allow=ulaw *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Jeremy Mann *Sent:* Monday, April 14, 2008 14:39 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Zap Codec Is there a way to force Zap channels to only use ulaw, and not even attempt g729 negotiation? My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I'm not licensed for the codec on the asterisk box. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization
Re: [asterisk-users] Zap Codec
Sadly you are correct: -- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0, _SIP_CODEC=ulaw) in new stack -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/156-083514c0, 4) in new stack -- Executing [EMAIL PROTECTED]:6] NoOp(SIP/156-083514c0, ) in new stack -- Executing [EMAIL PROTECTED]:7] Dial(SIP/156-083514c0, ) in new stack [Apr 15 09:51:25] WARNING[17434]: channel.c:3265 ast_request: No translator path exists for channel type Zap (native 76) to 256 [Apr 15 09:51:25] WARNING[17434]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:8] Hangup(SIP/156-083514c0, ) in new stack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec That would work just spiffy if you are calling another SIP device, but by the time the call gets to that point in the dialplan the codec of the originating device has already been chosen and set in stone. Tilghman Lesher wrote: On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. Set(_SIP_CODEC=ulaw) Dial(Zap/g0/...) -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Codec
I guess that's my frustration, I don't want it g729, I want it ulaw, I just wish Zap did codec negotiation from the client. It'd be a nice option instead of automatically trying to translate if it's not ulaw. Could save some processor overhead(obviously at the expense of bandwidth). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec If you want to get a G729 call to go via Zap you must purchase a G729 license. No amount of discussion is going to change that. Jeremy Mann wrote: Sadly you are correct: -- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0, _SIP_CODEC=ulaw) in new stack -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/156-083514c0, 4) in new stack -- Executing [EMAIL PROTECTED]:6] NoOp(SIP/156-083514c0, ) in new stack -- Executing [EMAIL PROTECTED]:7] Dial(SIP/156-083514c0, ) in new stack [Apr 15 09:51:25] WARNING[17434]: channel.c:3265 ast_request: No translator path exists for channel type Zap (native 76) to 256 [Apr 15 09:51:25] WARNING[17434]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:8] Hangup(SIP/156-083514c0, ) in new stack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec That would work just spiffy if you are calling another SIP device, but by the time the call gets to that point in the dialplan the codec of the originating device has already been chosen and set in stone. Tilghman Lesher wrote: On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. Set(_SIP_CODEC=ulaw) Dial(Zap/g0/...) -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided
Re: [asterisk-users] Zap Codec
Correct, but if I have two sip peers, one with G729ulaw, the other with gsmulaw, they will negotiate before trying to send audio. With ZAP, it tries to transcode whatever it receives into ulaw, period. No negotiation to even tell the client to send ulaw if capable. With no call level control(or dialplan logic, or anything!), I either use ulaw for ALL CALLS from sip peers(to other sip peers, to iax peers, to ZAP peers/channels), or use a combination of codecs and make sure it's able to be transcoded for the ZAP channels. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin Sent: Tuesday, April 15, 2008 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec Asterisk builds two channels and bridges them together. If the codecs mis-match then it must transcode, the negotiation on the Zap end is done seperately from the SIP end, so it does not care what your handset decided on. If you want ulaw, use ulaw, not g729 (on any call leg). You won't be able to mix and match codecs between calls, choose one for all calls and stick with it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Tuesday, April 15, 2008 08:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec I guess that's my frustration, I don't want it g729, I want it ulaw, I just wish Zap did codec negotiation from the client. It'd be a nice option instead of automatically trying to translate if it's not ulaw. Could save some processor overhead(obviously at the expense of bandwidth). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec If you want to get a G729 call to go via Zap you must purchase a G729 license. No amount of discussion is going to change that. Jeremy Mann wrote: Sadly you are correct: -- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0, _SIP_CODEC=ulaw) in new stack -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/156-083514c0, 4) in new stack -- Executing [EMAIL PROTECTED]:6] NoOp(SIP/156-083514c0, ) in new stack -- Executing [EMAIL PROTECTED]:7] Dial(SIP/156-083514c0, ) in new stack [Apr 15 09:51:25] WARNING[17434]: channel.c:3265 ast_request: No translator path exists for channel type Zap (native 76) to 256 [Apr 15 09:51:25] WARNING[17434]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:8] Hangup(SIP/156-083514c0, ) in new stack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec That would work just spiffy if you are calling another SIP device, but by the time the call gets to that point in the dialplan the codec of the originating device has already been chosen and set in stone. Tilghman Lesher wrote: On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. Set(_SIP_CODEC=ulaw) Dial(Zap/g0/...) -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com
[asterisk-users] Zap Codec
Is there a way to force Zap channels to only use ulaw, and not even attempt g729 negotiation? My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I'm not licensed for the codec on the asterisk box. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to give user a prompt before connecting thecall
I haven't, didn't know if you knew off the top of your head.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Tuesday, April 01, 2008 7:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to give user a prompt before connecting
thecall
I don't entirely remember - I was writing this code from memory.
Have you done any testing?
PaulH
On Tue, 2008-04-01 at 08:47 -0500, Jeremy Mann wrote:
Can I assume after exten=2,1,Playback(thanksfortakingthecall) there's more
logic, or does asterisk handle the connection between both parties at that
point?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Monday, March 31, 2008 9:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to give user a prompt before connecting
thecall
Something like this:
Dialling:
exten = s,n(dial),Dial($ZAP/G1/${number},15,M(check)gm)
exten = s,n,Dbget(next/number)
exten = s,n,Goto(dial)
{macro-check}
exten = s,n,Playback(${heresacall})
exten = s,n,Read(response,options,1)
exten = s,n,Goto(${response},1)
exten = 1,1,Macroexit
exten = 2,1,Playback(thanksfortakingthecall)
This hasn't been tested. Give it a red hot go.
Another option is to set up a queue with external numbers as members,
and set the queue as need the memebrs to accept the calls. (not that I
can remember that option)
PaulH
On Mon, 2008-03-31 at 20:55 -0500, Jeremy Mann wrote:
Please do!
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Paul Hales [EMAIL
PROTECTED]
Sent: Monday, March 31, 2008 7:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to give user a prompt before connecting
thecall
It can be done via the 'visit a macro' part of the dial command...
If anyone would like, i can post a code sample.
PaulH
On Mon, 2008-03-31 at 15:33 -0400, Dean Collins wrote:
Yes it is.
I'm remote at the moment so I can't send you the code but google for
mobile remote receiver and you'll find what you are looking for.
Lots of people do it so they don't have calls to cell phones picked up by
voicemail.
Cheers
dean
-Original Message-
From: Pete Kay [EMAIL PROTECTED]
Sent: Monday, March 31, 2008 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] How to give user a prompt before connecting
thecall
Hello,
Is it possible to request for the premission from the called party
through
a prompt before routing the call?
For instance, before actually connecting two parties through the use of
DIAL
command in the dialplan, I want to let Asterisk to automatically
ask for the called party to decide whether he/she would like to be
connected. ( ex. Press 1 to connect and 2 to hangup).
Can this function be done? If so, how to do it?
Thank you .
Pete Dao
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entity(ies) to whom it is addressed. If you are the intended recipient,
further disclosures are prohibited without proper authorization. If you are
not the intended recipient, any disclosure, copying, printing, or use of
this information is strictly prohibited and possibly a violation of federal
or state law and regulations. If you have received this information in
error, please notify Texas Health Management Group immediately at
1-817-310-4999. Texas Health Management Group, its subsidiaries, and
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information.
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Re: [asterisk-users] How to give user a prompt before connecting thecall
Can I assume after exten=2,1,Playback(thanksfortakingthecall) there's more
logic, or does asterisk handle the connection between both parties at that
point?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Monday, March 31, 2008 9:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to give user a prompt before connecting
thecall
Something like this:
Dialling:
exten = s,n(dial),Dial($ZAP/G1/${number},15,M(check)gm)
exten = s,n,Dbget(next/number)
exten = s,n,Goto(dial)
{macro-check}
exten = s,n,Playback(${heresacall})
exten = s,n,Read(response,options,1)
exten = s,n,Goto(${response},1)
exten = 1,1,Macroexit
exten = 2,1,Playback(thanksfortakingthecall)
This hasn't been tested. Give it a red hot go.
Another option is to set up a queue with external numbers as members,
and set the queue as need the memebrs to accept the calls. (not that I
can remember that option)
PaulH
On Mon, 2008-03-31 at 20:55 -0500, Jeremy Mann wrote:
Please do!
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Paul Hales [EMAIL
PROTECTED]
Sent: Monday, March 31, 2008 7:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to give user a prompt before connecting
thecall
It can be done via the 'visit a macro' part of the dial command...
If anyone would like, i can post a code sample.
PaulH
On Mon, 2008-03-31 at 15:33 -0400, Dean Collins wrote:
Yes it is.
I'm remote at the moment so I can't send you the code but google for mobile
remote receiver and you'll find what you are looking for.
Lots of people do it so they don't have calls to cell phones picked up by
voicemail.
Cheers
dean
-Original Message-
From: Pete Kay [EMAIL PROTECTED]
Sent: Monday, March 31, 2008 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] How to give user a prompt before connecting
thecall
Hello,
Is it possible to request for the premission from the called party through
a prompt before routing the call?
For instance, before actually connecting two parties through the use of DIAL
command in the dialplan, I want to let Asterisk to automatically
ask for the called party to decide whether he/she would like to be
connected. ( ex. Press 1 to connect and 2 to hangup).
Can this function be done? If so, how to do it?
Thank you .
Pete Dao
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This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and entity(ies)
to whom it is addressed. If you are the intended recipient, further
disclosures are prohibited without proper authorization. If you are not the
intended recipient, any disclosure, copying, printing, or use of this
information is strictly prohibited and possibly a violation of federal or
state law and regulations. If you have received this information in error,
please notify Texas Health Management Group immediately at 1-817-310-4999.
Texas Health Management Group, its subsidiaries, and affiliates hereby claim
all applicable privileges related to this information.
___
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prohibited and possibly a violation of federal or state law and regulations. If
you have
Re: [asterisk-users] How to give user a prompt before connecting thecall
Please do! From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Paul Hales [EMAIL PROTECTED] Sent: Monday, March 31, 2008 7:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to give user a prompt before connecting thecall It can be done via the 'visit a macro' part of the dial command... If anyone would like, i can post a code sample. PaulH On Mon, 2008-03-31 at 15:33 -0400, Dean Collins wrote: Yes it is. I'm remote at the moment so I can't send you the code but google for mobile remote receiver and you'll find what you are looking for. Lots of people do it so they don't have calls to cell phones picked up by voicemail. Cheers dean -Original Message- From: Pete Kay [EMAIL PROTECTED] Sent: Monday, March 31, 2008 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] How to give user a prompt before connecting thecall Hello, Is it possible to request for the premission from the called party through a prompt before routing the call? For instance, before actually connecting two parties through the use of DIAL command in the dialplan, I want to let Asterisk to automatically ask for the called party to decide whether he/she would like to be connected. ( ex. Press 1 to connect and 2 to hangup). Can this function be done? If so, how to do it? Thank you . Pete Dao ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplan Help
I've got a couple of extensions in users.conf that have both SIP and IAX
access(IAX softphone, SIP hard phone).
I'd like to setup my dial string to check to see which they are actively
registered with, and send the call appropriately.
Right now I have:
Exten = _4xx,1,Dial(SIP/${EXTEN}IAX2/${EXTEN})
But not all phones have both techs, so there is a lot of misses
Is there a way to use the hints to see which they are registered with, and dial
only using those channel types?
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[asterisk-users] DUNDi
Is there a way to have a dundi host advertise extensions for another server? A---B---C I'd like A to reach C through B. A and C would handle the call, B would just be the DUNDi intermediary. Assuming A has 101-199 B has 201-299 And C has 301-399 A sample dundi/extensions/iax config for B is all I need. I can get single DUNDi queries running fine(A-B, B-C, A-C(directly setup in dundi.conf)) I just have 25 offices that all connect to a central location, I'd rather the central location be the hub of all dundi queries for all other locations. Thanks. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi
Nevermind, figured it out. I had restrictions on the unsolicited calls in dundi.conf. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Wednesday, March 12, 2008 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DUNDi Is there a way to have a dundi host advertise extensions for another server? A---B---C I'd like A to reach C through B. A and C would handle the call, B would just be the DUNDi intermediary. Assuming A has 101-199 B has 201-299 And C has 301-399 A sample dundi/extensions/iax config for B is all I need. I can get single DUNDi queries running fine(A-B, B-C, A-C(directly setup in dundi.conf)) I just have 25 offices that all connect to a central location, I'd rather the central location be the hub of all dundi queries for all other locations. Thanks. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe Admin Functions
Is there any way that I can have an admin user hit * and then Mute all other users in a meetme conference? Sort of a moderator function? I know it can be done with MeetMeAdmin, but as I see it that requires a separate extension to dial, unless I've got the logic wrong? If it can be done in a single extension please show examples. Thanks. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extension Logic Help
To you extensions.conf gurus, I'd like some help on having a button/feature to turn on/off system wide call forwarding. I need the phone system to forward calls received, after the feature is activated, to an answering service. Calls received are on a PRI. I need all DIDs forwarded once the feature is activated. The forwarding will go out the same PRI, and ideally CallerID will be passed through. I don't anticipate more than 2 calls simultaneous. Thanks. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Admin Functions
Perfect! Thanks.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield
Sent: Tuesday, February 19, 2008 11:01 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MeetMe Admin Functions
In article [EMAIL PROTECTED],
Jeremy Mann [EMAIL PROTECTED] wrote:
Is there any way that I can have an admin user hit * and then Mute all other
users in a
meetme conference? Sort of a moderator function?
I know it can be done with MeetMeAdmin, but as I see it that requires a
separate extension
to dial, unless I've got the logic wrong?
If it can be done in a single extension please show examples.
Use the X option in MeetMe to allow a single digit to be entered which
will exit the conference and go to that extension number. This is used
without pressing * first.
At that extension you can execute MeetMeAdmin on the same conference to
mute all the non-admin users, and then execute MeetMe again to go straight
back into the conference. e.g.
[conf]
; conference must be defined or saved in channel variable CONF
; note that _X. matches 2 digits or more,
; leaving single digit exts available
exten = _X.,1,NoOp(entering conference ${CONF})
exten = _X.,n,Set(SAVEDEXTEN=${EXTEN})
exten = _X.,n,MeetMe(${CONF},X)
exten = _X.,n,Hangup
; allow user to press 5 to mute all users
exten = 5,1,NoOp(muting conference ${CONF})
exten = 5,n,MeetMeAdmin(${CONF},N)
exten = 5,n,Goto(${SAVEDEXTEN},1)
; allow user to press 6 to unmute all users
exten = 6,1,NoOp(unmuting conference ${CONF})
exten = 6,n,MeetMeAdmin(${CONF},n)
exten = 6,n,Goto(${SAVEDEXTEN},1)
Hope this helps!
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] modem through Zaptel/Digium?
Is it bridging the Zap channels? We have asterisk doing FXO-FXS modem calls working fine, the key is making sure the channels are bridging and EC is NOT turning on. If you have anything preventing that the modem calls won't work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Thursday, January 17, 2008 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] modem through Zaptel/Digium? Greg Woods wrote: This is just a low priority curiosity question because I have a usable workaround. I have Digium card that uses the Zaptel driver (can't get to my home machine right now to get the exact model, but it probably doesn't matter). It's a card with one POTS line and three extension hookups. I'm using Asterisk 1.4 and Zaptel 1.4.7 . One of the extension ports is connected to a modem on another computer. This is a FAX modem that works well; I have * programmed to detect incoming faxes and route them to this modem, and it works seamlessly. I can also send outbound faxes with no problem. The curiosity is that this modem does not work for dialup unless I bypass the * server and connect it directly to the wallplate, then it works fine. I don't see why it would be able to detect carrier and negotiate with a fax machine through * and Zaptel, but not with a dialup server. --Greg I think asterisk has the ability to detect fax tones and disable echo cancellation for those calls. I don't know if that is the case with a regular modem call. I'd check to make sure that echo cancellation is disabled on the extension the modem is plugged into. The only other idea is to try connecting at a lower speed (I would think this would happen automatically though). -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap Issues
Using Asterisk-1.4.17, Zaptel-1.4.8, libpri-1.4.3 Upgraded this morning, now PRI channels are unstable as hell. After about 5 minutes all asterisk commands on the console refuse to respond, attached is the debug log right before and after the lock-up, IT occurred between 9:18 and 9:20 AM at 9:20 I restarted asterisk. Box is debian w/ asterisk built from scratch. My setup is asterisk as a man-in-the-middle, Span 1 goes to Telco, Span 2 to Nortel MICS. PRI is not the problem as it's plugged into the Nortel directly for now and we have no problems. Nothing in dmesg indicates any errors. Any clue how I go about debugging this? [Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Unlinking slave 1 from 47 [Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Removed 12 from conference 9/47 [Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Removed 57 from conference 9/1 [Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1 [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Unlinking slave 26 from 3 [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Removed 36 from conference 9/3 [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Removed 14 from conference 9/26 [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/26-1 [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/26-1 [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/3-1 [Jan 16 09:20:24] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann el. [Jan 16 09:20:24] DEBUG[8430] chan_zap.c: Found empty available channel 0/21 [Jan 16 09:22:24] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann el. [Jan 16 09:22:24] DEBUG[8430] chan_zap.c: Found empty available channel 0/20 [Jan 16 09:22:31] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann el. [Jan 16 09:22:31] DEBUG[8430] chan_zap.c: Found empty available channel 0/19 [Jan 16 09:23:07] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann el. [Jan 16 09:23:07] DEBUG[8430] chan_zap.c: Found empty available channel 0/18 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Heartbeat
Has anyone ever written asterisk logic to Heartbeat remote phone lines? Something that would dial out and see if a busy tone is encountered and take some sort of action? If not, any good ideas on how to do it? Obviously this would involve .call files. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom VLAN
Just curious, if I have my Polycom IP 550 phone VLAN tag 30, will the packets I send from my PC(on the PC port of the phone) have the same VLAN tag? THe PC is sending untagged packets. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Aastra 480i CT
Are the cordless phones on the 480i CT from Aastra registered independently in Asterisk? Such that if I have 5 of the cordless phones hooked up, each one is it's own extension? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma Question
Do sangoma cards use the standard Zaptel drivers? Or do they have to be compiled externally like Rhino cards? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Question
And they work with Asterisk/Zaptel 1.4 ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Wednesday, November 28, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma Question On Nov 28, 2007 10:52 AM, Jeremy Mann [EMAIL PROTECTED] wrote: Do sangoma cards use the standard Zaptel drivers? Or do they have to be compiled externally like Rhino cards? Sangoma maintains a patchset that gets applied to the stock zaptel drivers before compilation. They provide automated tools that will take care of the patching/compiling/installing/configuring for you. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail issues in 1.4.11
That was exactly it. Default 1.4 install include unavail.ulaw, which was matching over all other recordings. When I deleted the useless files it went fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Tuesday, October 16, 2007 8:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail issues in 1.4.11 Jeremy Mann wrote: Asterisk isn't playing my voicemail greetings even though they are defined. Below are the relevant configs(from show dialplan) as well as the level 3 verbose messages asterisk is giving. Also a listing of the directory. Asterisk just plays the The person at extension... message, not the greetings I have recorded. Try to get rid of all of the unavail files in your directory but one. I had the same thing happen to me and realized that I had lost track of which files were valid and which weren't (IOW, I had a bunch of empty or corrupt audio files). Use the process of elimination to find a file and codec that works. You might also try doing a stack trace if increasing the verbosity doesn't help you find the problem. -Stephen- This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail issues in 1.4.11
Asterisk isn't playing my voicemail greetings even though they are defined.
Below are the relevant configs(from show dialplan) as well as the level 3
verbose messages asterisk is giving. Also a listing of the directory.
Asterisk just plays the The person at extension... message, not the greetings
I have recorded.
Thanks
--
asterisk*CLI show dialplan macro-stdexten
[ Context 'macro-stdexten' created by 'pbx_config' ]
'a' =1. VoicemailMain(${ARG1}) [pbx_config]
's' =1. Dial(${ARG2}|30) [pbx_config]
2. Goto(s-${DIALSTATUS}|1)[pbx_config]
's-BUSY' = 1. Voicemail(${ARG1}|b) [pbx_config]
2. Goto(default|s|1) [pbx_config]
's-NOANSWER' = 1. Voicemail(${ARG1}|u) [pbx_config]
2. Goto(default|s|1) [pbx_config]
'_s-.' = 1. Goto(s-NOANSWER|1) [pbx_config]
-= 5 extensions (8 priorities) in 1 context. =-
asterisk*CLI show dialplan internal
[ Context 'internal' created by 'pbx_config' ]
'_1234' =1. Macro(stdexten|[EMAIL PROTECTED]|SIP/1234)
[pbx_config]
2. Hangup() [pbx_config]
-- Nobody picked up in 3 ms
-- Executing [EMAIL PROTECTED]:2] Goto(Zap/44-1, s-NOANSWER|1) in new
stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [EMAIL PROTECTED]:1] VoiceMail(Zap/44-1, [EMAIL
PROTECTED]|u) in new stack
-- Zap/44-1 Playing '/var/spool/asterisk/voicemail/default/1234/unavail'
(language 'en')
-- Zap/44-1 Playing 'vm-intro' (language 'en')
[EMAIL PROTECTED]:/usr/src/asterisk-1.4.11/configs# ls
/var/spool/asterisk/voicemail/default/1234
busy busy.g729 busy.ulaw busy.WAV greet.wav INBOX temp unavail
unavail.g729 unavail.ulaw unavail.WAV
busy.alaw busy.gsm busy.wav greet.gsm greet.WAV Oldtmp
unavail.alaw unavail.gsm unavail.wav
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Re: [asterisk-users] Replace full PRI with SIP/IAX trunks...YES/NO?
Without knowing more, Why fix what isn't broken? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Canfield Sent: Friday, October 05, 2007 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Replace full PRI with SIP/IAX trunks...YES/NO? I've been considering replacing a PRI with SIP or IAX trunks. The monthly cost difference is marginal, but it would save a bit on the hardware side and soft trunks would be easier to manage. I can't help but wonder what I would be giving up? I'd like to hear some lessons learned from those who are doing it or decided, for whatever reason, it's a bad idea. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replace full PRI with SIP/IAX trunks...YES/NO?
Get a 2 port card, problem solved. Asterisk is the Man-in-the-Middle. I'm running this right now between an asterisk box and Nortel MICS system. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Canfield Sent: Friday, October 05, 2007 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Replace full PRI with SIP/IAX trunks...YES/NO? Jeremy Mann wrote: Without knowing more, Why fix what isn't broken? I should have stated, the PRI is on an existing PBX not asterisk. My goal was to reuse the existing PBX PRI card to interface with asterisk. I've been considering replacing a PRI with SIP or IAX trunks. The monthly cost difference is marginal, but it would save a bit on the hardware side and soft trunks would be easier to manage. I can't help but wonder what I would be giving up? I'd like to hear some lessons learned from those who are doing it or decided, for whatever reason, it's a bad idea. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rhino RCB8FXX
Anyone know if Rhino is planning on supporting zaptel 1.4 anytime soon? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rhino RCB8FXX
Latest being 1.1.1 ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Tuesday, October 02, 2007 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Rhino RCB8FXX
We are using it successfully with zaptel 1.4 -- just be sure and get
the latest drivers which are now independent of the zaptel sources.
on Tuesday 10/02/2007 Jeremy Mann([EMAIL PROTECTED]) wrote
Anyone know if Rhino is planning on supporting zaptel 1.4 anytime soon?
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p class=MsoNormalAnyone know if Rhino is planning on supporting zaptel
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[asterisk-users] Multiple Home system with SIP
Is there a way to tell asterisk, via a sip.conf peer, what IP address to send a packet out of? I've got multiple NICs in my box, each with it's own public IP. I need the SIP messages to originate from any one of the IPs depending on which number was originally called(and therefore where the packet originally came from). My fear is that it will listen on all IPs fine, but only respond via the default GW. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Home system with SIP
Why did you waste time with this reply? You do realize some users don't have control over their Exchange servers, and asinine footers are placed into an email without their intervention or control right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benny Amorsen Sent: Tuesday, September 25, 2007 1:55 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple Home system with SIP JM == Jeremy Mann [EMAIL PROTECTED] writes: I would have answered, but I was prohibited from quoting properly: JM If you are the intended recipient, further disclosures are JM prohibited without proper authorization. /Benny This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Home system with SIP
And if the Sip provider is sending data from 1 or two fixed hosts? For instance, they send DID1 to IP A.B.C.D from 1.1.1.1 They send DID2 to IP E.F.G.H from 1.1.1.1 How do you differentiate? Would fromhost= work? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Question
I'm curious if anyone has implemented the following: Need to setup an on-call queue, that activates after 5PM and de-activates at 8AM, also that activates/deactivates on demand(I'm thinking a feature code here). The agents need to log in via cell phones, and when calls come in from outside to the asterisk system, it'll need to call the cell phone agents that are active. I'm thinking that it's a simple SQL query, to update the agents status and number, and that asterisk will do a lookup and append that to the ZAP channel to dial, but interested in any logic someone might be able to come up with for the dialplan. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
Does G.729 phone - asterisk - G.729 phone work with reinvite turned off? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Watson Sent: Tuesday, September 18, 2007 1:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 PSTN - g729 requires transcoding at that point. You can however do: G.729 phone - asterisk - G.729 phone without license (from my understanding). But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it requires a license to preform transcoding. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman Sent: September-18-07 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Failover SIP logic
I need some extensions logic assistance, I'm trying to dial out one of multiple
SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call
per trunk) and roll over to a second or third depending on that busy status
Here's what I've got for a macro thusfar, but it's not working(fails if the 1st
trunk is busy)
extensions.conf:
[globals]
trunk_1 = SIP/trunk1
trunk_2 = SIP/trunk2
trunk_3 = SIP/trunk3
[macro-trunkdial]
exten = s,1,Dial(${trunk_1}/${ARG1})
exten = s,2,Hangup()
exten = s,102,Dial(${trunk_2}/${ARG1})
exten = s,103,Hangup()
exten = s,203,Dial(${trunk_3}/${ARG1})
exten = s,204,Hangup()
[from-internal]
exten = _NXXNXX,1,Macro(trunkdial,+1${EXTEN})
exten = _1NXXNXX,1,Macro(trunkdial,+${EXTEN})
sip.conf:
[trunk1]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
[trunk2]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
[trunk3]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
Here's asterisk output when someone dials out:
Executing [EMAIL PROTECTED]:1] Macro(SIP/6001-007e2840,
trunkdial|+1xx) in new stack
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/6001-007e2840,
SIP/trunk1/+1xx) in new stack
[Sep 10 09:06:52] ERROR[16253]: chan_sip.c:3192 update_call_counter: Call to
peer 'trunk1' rejected due to usage limit of 1
-- Couldn't call trunk1/+1xx
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/6001-007e2840, ) in new
stack
I don't want the dialplan to cascade like:
exten = 1,dial...
exten = 2,dial...
Because if the remote end hangs up I don't want it going to priority 2 to dial
out again(in case my user doesn't hit hang-up on their end) so I need logic to
detect a busy channel and jump to the next section..
Thanks for any help.
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Re: [asterisk-users] Failover SIP logic
Asterisk 1.4.11
Sorry, meant to include that
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrea Spadaccini
Sent: Monday, September 10, 2007 10:59 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Failover SIP logic
Ciao Jeremy,
I need some extensions logic assistance, I'm trying to dial out one of
multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only
allow 1 call per trunk) and roll over to a second or third depending on that
busy status
Here's what I've got for a macro thusfar, but it's not working(fails if the
1st trunk is busy) extensions.conf:
[globals]
trunk_1 = SIP/trunk1
trunk_2 = SIP/trunk2
trunk_3 = SIP/trunk3
[macro-trunkdial]
exten = s,1,Dial(${trunk_1}/${ARG1})
exten = s,2,Hangup()
exten = s,102,Dial(${trunk_2}/${ARG1})
exten = s,103,Hangup()
exten = s,203,Dial(${trunk_3}/${ARG1})
exten = s,204,Hangup()
Which asterisk version are you using?
IIRC, priority jumping (ie. going to n+101) was disabled by default in some
1.2.x version. You should rely on DIALSTATUS. See Dial() page in voip-info.org.
HTH,
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any disclosure, copying, printing, or use of this information is strictly
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you have received this information in error, please notify Texas Health
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[asterisk-users] Cisco UC 500
Is the Cisco UC 500 able to integrate with Asterisk? Specifically does it work via SIP? Just curious, as the Cold Call Cisco sales rep had no clue what SIP even was, and this device looks interesting. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inbound SIP issues
I have an issue with receiving inbound calls. I've got bandwidth.com trunks incoming to my asterisk box, bandwidth sends all incoming traffic to one of two IP addresses, and requires outbound traffic go to either of the same two IP addresses. I've got to use fromuser=DID on outgoing calls so they apply the right caller ID. My issue is that I want incoming calls to match on a specific sip.conf entry, but they are matching on my outgoing entries and dropping(I don't have context associated with them). Here's relevant sip.conf entries -- [bandwidth_inbound_1] host=4.79.212.236 port=5060 type=peer disallow=all allow=ulaw dtmfmode=rfc2833 canreinvite=no nat=no context=frombandwidth [bandwidth_inbound_2] host=216.82.224.202 port=5060 type=peer disallow=all allow=ulaw dtmfmode=rfc2833 canreinvite=no nat=no context=frombandwidth [bandwidth_outbound_did1] host=4.79.212.236 port=5060 type=peer disallow=all allow=ulaw dtmfmode=rfc2833 canreinvite=no reinvite=no nat=no fromuser=did1 If calls come in from 4.79.212.236 they are immediately matched to context [bandwidth_outbound_did1] If I put the inbound contexts under the outbound in sip.conf they work, is that the design intention of sip.conf? Bandwidth doesn't require or accept register statements, so I can't use that to send calls to specific extensions. Is there any easier logic to attach my fromuser when I have multiple DIDs? Ideally I'd love 2 entries for them total. I'm running asterisk 1.4.11 if it helps. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Question
I have a SIP phone calling via a SIP trunk another asterisk system, that then
sends the call out a ZAP channel.
When I press any of the features defined in features.conf, The end user on the
ZAP side hears the DTMF tones, and none of the features work.
My DTMFmode on the SIP users definition is rfc2833
Asterisk console doesn't register that a feature is being recognized, any ideas?
Below are my users.conf definition, features.conf, and extensions logic on the
system the SIP caller is registered with(users.conf setup with AsteriskGUI,
extensions.conf setup by hand):
--features.conf
[featuremap]
blindxfer = #1 ; Blind transfer (default is #)
disconnect = *0; Disconnect (default is *)
automon = *1 ; One Touch Record a.k.a. Touch Monitor
atxfer = *2; Attended transfer
parkcall = #72; Park call (one step parking)
--users.conf--
[6003]
callwaiting = no
context = from-internal
fullname = IT Support
hasagent = no
hasdirectory = no
hasiax = yes
hasmanager = no
hassip = yes
hasvoicemail = no
host = dynamic
mailbox = 6003
threewaycalling = no
vmsecret = 1234
registeriax = yes
registersip = yes
canreinvite = no
nat = no
dtmfmode = rfc2833
disallow = all
allow = all
[dtrr] ;bi-directional trunk to 2nd asterisk system.
allow = ulaw,alaw
context = from-outside
dialformat = ${EXTEN}
hasexten = no
hasiax = no
hassip = yes
host = 10.10.0.10
port = 5060
username = dtrr
secret = dtrr
registeriax = no
registersip = no
trunkname = Custom - Corporate
trunkstyle = customvoip
disallow = gsm,ilbc,speex,g726,adpcm,lpc10,g729
--extensions.conf--
[globals]
trunk_2 = SIP/dtrr
[macro-trunkdial]
exten = s,1,Dial(${ARG1})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,Hangup()
exten = s-NOANSWER,1,Hangup()
exten = _s-.,1,NoOp()
[from-internal]
exten = _1000,1,Macro(trunkdial,${trunk_2}/${EXTEN})
exten = _1NXXNXX,1,Macro(trunkdial,${trunk_2}/${EXTEN})
exten = _NXXNXX,1,Macro(trunkdial,${trunk_2}/${EXTEN})
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[asterisk-users] AsteriskNOW Web GUI
Is the web GUI for AsteriskNOW able to be loaded on an existing server(that was installed from ubuntu-server and asterisk loaded from source)? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI Question
For 1.4: core set verbose 2 For 1.2: set verbose 2 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Andersen Sent: Tuesday, August 21, 2007 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CLI Question When I use the CLI (asterisk -r) I get all sorts of info scrolling past about current activity such as... -- Executing Macro(SIP/7110-b1d316e0, callrecord|7134) in new stack -- Executing NoOp(SIP/7110-b1d316e0, Call Record Macro REC7134 ) in new stack -- Executing GotoIf(SIP/7110-b1d316e0, 0?4:3) in new stack -- Goto (macro-callrecord,s,3) -- Executing GotoIf(SIP/7110-b1d316e0, 0?4:8) in new stack -- Goto (macro-callrecord,s,8) -- Executing NoOp(SIP/7110-b1d316e0, ) in new stack -- Executing GotoIf(SIP/7110-b1d316e0, 1?8:150) in new stack -- Goto (macro-simpleexten,s,8) and soforth... I'm trying to learn the CLI and so I type something like: sip showtabtab and I get a list of other options. BUT, before I get through reading what is on the screen, a call comes and and scrolls up the screen with the info above. Is there a flag to pass to rasterisk to tell it only show info related to my queries and don't keep showing me all the current call status? (less verbose?) Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback
1. Yes 2. Yes 3. Yes Nice sales pitch, sounds like one of those late night get rich now! schemes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Friday, August 17, 2007 4:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? 2. Does the complexity of the DUNDi setup discourage you from using it or even attempting to configure it? 3. If there was a simple tutorial, step by step guide with easy to setup and test examples, would this encourage more users to investigate and use DUNDi? I'm interested in putting together a new-user tutorial about DUNDi configuration and setup. There is a lot of great information, setup guides already but the feedback I get is that the current examples are a bit complicated to follow for new users. Your feedback is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX Trunk
Is there a way to limit IAX trunks to a certain number of calls? For instance, if I'm linking two systems in different regions, can I limit the number of calls that go across IAX between the systems? I've got some dialplan logic, but if there's some iax.conf directive to limit the number of calls it'd be so much simpler. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Question
Good idea! It's working great. I also like your local vs LD logic, much
simpler to do than NXXNXX or 1NXXNXX.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck
Sent: Tuesday, August 14, 2007 8:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI Question
You can eliminate the set CallerID line. This will just set the
variable back to itself. Asterisk will pass the callerid from one
span to the next.
You can use a GotoIF to set the callerid to something else if it is
blank or marked as Private:
exten = s,1,GoToIf($[${CALLERID(num)} = ]?2:3)
exten = s,2,Set(CALLERID(num)=00)
If the CallerID number is blank go to 2 else go to 3.
I wonder if asterisk or the norstar system is holdng on to that last
callerid number on the channel?
The only time you may want to set callerid is when your Norstar dials
out through Asterisk:
[norstar]
; This context is where all incoming calls from the norstar are placed
; Basically take the call from the norstar and bridge it over to the first
; available line on the bottom of the T1 going to TimeWarner.
exten = _1900XXX,1,Playback(cannot-complete-as-dialed)
exten = _1900XXX,2,Hangup()
exten = _1X.,1,GoToIf($[${CALLERID(num)} = ]?2:3)
exten = _1X.,2,Set(CALLERID(num)=511212)
exten = _1X.,3,NoOp(${CALLERID(num)})
exten = _1X.,4,Dial(${PRITRUNK}/${EXTEN},300,tD())
exten = _1X.,5,Hangup()
exten = _X.,1,GoToIf($[${CALLERID(num)} = ]?2:3)
exten = _X.,2,Set(CALLERID(num)=511212)
exten = _X.,3,NoOp(${CALLERID(num)})
exten = _X.,4,Dial(${PRITRUNK}/${EXTEN},300,)
exten = _X.,5,Hangup()
exten = i,1,Answer()
exten = i,n,Wait(1)
exten = i,n,Playback(cannot-complete-as-dialed)
exten = i,n,Playback(please-contact-tech-supt)
exten = i,n,Hangup()
On 8/9/07, Mike Lynchfield [EMAIL PROTECTED] wrote:
hmm from what i have seen this is not supposed to be.. the info is still
there but should not be used in case of privacy..
zap show channels always show last info till a span refresh.. but the
privacy should indeed replace those with Privacy.
Maybe it could be a bug ,
On 8/9/07, Jeremy Mann [EMAIL PROTECTED] wrote:
I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco,
Span 2 sends to my existing phone system(Nortel).
My Span1 gets sent to the context from-pri, detailed here:
[from-pri]
exten = _49XX,1,Set(CALLERID(all)=${CALLERID(all)})
exten = _49XX,2,Dial(Zap/g2/${EXTEN},,twk)
exten = _49XX,3,Congestion()
exten = _49XX,4,Set(CALLERID(all)=)
exten = _49XX,5,Hangup()
exten = _49XX,103,Congestion()
exten = _49XX,104,Set(CALLERID(all)=)
exten = _49XX,105,Hangup()
exten = h,1,Set(CALLERID(all)=)
exten = h,2,Hangup()
I'm receiving caller ID fine, and setting it on the outgoing channel the
same I received it, is my logic above wrong? Will Asterisk natively pass
through the caller ID, or is there a better way to set it?
The reason I ask, is that calls that are not coming in with CLID(blocked
or private) are showing up as the same number that was previously answered
on that channel.
Thanks.
Using Asterisk 1.4 FYI.
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not the intended recipient, any disclosure, copying, printing, or use of
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Making it happen
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1.514.312.7030
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[asterisk-users] DTMF on Bridged ZAP call
Should asterisk be intercepting DTMF on a bridged ZAP call? If so, how do I disable it recognizing #, as it's hanging up my users when they try to enter #. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recognize 800 number
Is there a way to recognize if someone called our PRI using an 800 number? The DID is showing my 4 digit primary line, not anything obvious signifying that an 800 number is called? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR-CSV Processing
Does anyone have any tools to process CDR-CSV files into reports? I don't have anything specific in mind, I'd just like some reporting examples so I don't have to reinvent the wheel. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR-CSV Processing
Every little bit helps, thanks! I guess I'm actually going to look at cost/benefit analysis, trying to see where calls are going across IAX and tallying up what would have been LD cost(we're doing intra-office IAX calling where possible) to tag as savings to justify the systems to our ownership. I really only care about answered calls, though I can imagine 5-10 reports about unanswered, abandonded, congestion, etc that would benefit us in the long term as well. Thanks for your sample. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Monday, August 13, 2007 12:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR-CSV Processing Free and inexpensive aren't quite the same. I don't follow the -biz list because I don't want to hear plugs from You at Evariste about this stuff. I sent a PHP snippet to the list maybe a year and a half ago, search something like site:lists.digium.com Mojo csv or site:lists.digium.com Mojo cdr or site:lists.digium.com Mojo php and see if you can find it, let me know if you can't. While it only totaled columns, it may be a nice starting point for you. Moj Alex Balashov wrote: We at Evariste have a lot of experience writing all sorts of custom CDR reports and would be happy to write what you need for you--very inexpensively, guaranteed. On Mon, 13 Aug 2007, Jeremy Mann wrote: Does anyone have any tools to process CDR-CSV files into reports? I don't have anything specific in mind, I'd just like some reporting examples so I don't have to reinvent the wheel. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI Question
I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco, Span
2 sends to my existing phone system(Nortel).
My Span1 gets sent to the context from-pri, detailed here:
[from-pri]
exten = _49XX,1,Set(CALLERID(all)=${CALLERID(all)})
exten = _49XX,2,Dial(Zap/g2/${EXTEN},,twk)
exten = _49XX,3,Congestion()
exten = _49XX,4,Set(CALLERID(all)=)
exten = _49XX,5,Hangup()
exten = _49XX,103,Congestion()
exten = _49XX,104,Set(CALLERID(all)=)
exten = _49XX,105,Hangup()
exten = h,1,Set(CALLERID(all)=)
exten = h,2,Hangup()
I'm receiving caller ID fine, and setting it on the outgoing channel the same I
received it, is my logic above wrong? Will Asterisk natively pass through the
caller ID, or is there a better way to set it?
The reason I ask, is that calls that are not coming in with CLID(blocked or
private) are showing up as the same number that was previously answered on that
channel.
Thanks.
Using Asterisk 1.4 FYI.
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
its subsidiaries, and affiliates hereby claim all applicable privileges related
to this information.
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[asterisk-users] Zap Bridge Question
asterisk*CLI show channels Channel Location State Application(Data) Zap/3-1 (None) Up Bridged Call(Zap/47-1) Zap/47-1 [EMAIL PROTECTED] Up Dial(ZAP/g1/2105||TWK) Zap/25-1 (None) Up Bridged Call(Zap/1-1) Zap/1-1 [EMAIL PROTECTED]:2 Up Dial(Zap/g2/4999||twk) Zap/26-1 (None) Up Bridged Call(Zap/2-1) Zap/2-1 [EMAIL PROTECTED]:2 Up Dial(Zap/g2/4999||twk) Can I assume those calls are truly bridged above? If so why does zap show channel show me the Echo Cancellation is active when I have requested it not be active on bridged calls? System is a 2x Digium T1 card, one connects to PSTN the other to a Nortel phone system. Zapata.conf follows, if I'm missing something to ensure zap channel bridging please let me know. [trunkgroups] [channels] language=en context=default switchtype=national rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes cidsignalling=bell hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 immediate=no switchtype=national signalling=pri_cpe context=from-pri channel=1-23 group=2 signalling=pri_net context=from-nortel channel=25-47 signalling=fxo_ks channel=49 signalling=fxs_ks channel=52 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI Reset
Is it normal for a PRI to reset the inactive B channels periodically(like once every hour). I'm seeing on my asterisk console successful restarts, just curious as this is all new to me. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE207P Question
I need help on my zaptel.conf and Zapata.conf for a TE207P I'd like Span 1 to receive a PRI from the phone company(US PRI). I'd like Span 2 to interface with a Nortel Phone system as a PRI(acting as the phone company) Essentially my asterisk box is a man in the middle intercepting calls from the PRI passing certain DID to the Nortel, also intercepting calls from the Nortel passing them via IAX to other asterisk boxes as necessary. Do I just need to make both PRI signaling? See below: /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 /etc/asterisk/Zapata.conf Group=1 Signaling=pri_cpe Switchtype=national Context=from-pri Channel=1-23 Group=2 Context=from-nortel Channel=25-47 Thanks for any help. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE207P Question
So would the timing be 0? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Tuesday, August 07, 2007 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TE207P Question As an added note, you may want to change the timing setting on your second span= line in zaptel.conf. If you're acting as the telco, you might want to send timing to your Nortel, depending on how the Nortel is configured. (The way you currently have your spans configured, you're telling Zaptel to get it's timing from both the telco and the Nortel, and the card can only sync to one timing source at a time.) This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Switchtype
In Zapata.conf, if my PRI is NI-2 configured, do I still use switchtype=national ? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] USB Cordless
Does anyone know if X-Ten or SJPhone support multiple cordless handsets for multiple lines? I have an office with multiple roaming users(nurses) that are in and out. I'd like to provide them telephones, and my idea is to have a PC sitting in a corner somewhere running a softphone client. When a nurse comes in she just picks up any available handset(anywhere from 2-5 per office) and starts calling. Each handset would be labeled with their extension so that if any inbound calls came to them they'd be able to let the receptionist know their extension. Any ideas? Also, is it possible to transfer a call directly to someone's VM(if they are out of the office) bypassing their extension? If so, could someone post the asterisk logic behind the extension setup? I don't want anything too complex(like setting the DND or phone to busy). Thanks. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
you would think the telcos would be more interested in selling this to small/medium businesses that are not ready for a voice pri but it Since when to the telcos have the consumer's best interest in mind? They can sell you a PRI at full loop cost with a smaller number of channels in the hopes you will add to it, they will then charge you an upgrade fee or some other inflated installation cost when in reality it is almost 0 work to reprovision, pure profit for them. ATT is/was doing buyback promotions recently for 5 analog lines + a full Data T1 for around $425 total(including loop cost), that's a steal and frankly we would have been crazy to request BRI service. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ESI Phone System Integration
ESI Phone systems are supposed to support IP stations via SIP integration(http://www.esi-estech.com/products/systems/ESICS/), has anyone ever tried to link Asterisk with one of these? I'm thinking my asterisk box could be an extension off that phone system, that would then provide a Dial by Name directory to use. Not elegant, but it'd work. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- 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] Using Modems with Asterisk
Do you just passthrough from FXO to FXS on the channel bank? Does asterisk do the passthrough or the channel bank itself? I ask because we're considering an Adit 600 internally and that's one of my pending questions. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, June 13, 2007 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Modems with Asterisk Lutgring, Sam wrote: Has anyone had any experience using a modem through the Asterisk system? I have some technical support personnel that need to use a computer modem to connect to a remote system for troubleshooting. Is there a SIP compliant This will probably not work for the same reason faxes won't work. We use a Adit 600 channel bank for modem communications. 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- 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] Using Modems with Asterisk
So you're doing PRI-Channel bank? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, June 13, 2007 11:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Modems with Asterisk Jeremy Mann wrote: Do you just passthrough from FXO to FXS on the channel bank? Does asterisk do the passthrough or the channel bank itself? The Adit hooks up to the Asterisk via a T1 cable, so you'd need a Dual PRI card in your Asterisk box. Our channel bank is on channels 25-48. Asterisk handles the routing. 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- 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
[asterisk-users] Integrated T1
Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? It's only going to support 4-5 users(the voice channels won't all be active obviously). This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- 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] FXS + Pots Extensions Help
Here's a silly question, if these are standard POTS you obviously know which number corresponds to which line, being the case couldn't you tell that ZAP/1 is POTS 555-1234, ZAP/2 is POTS 555-1235, etc etc? I'm assuming you're trying to identify the inbound number from the telco that was dialed. Unless you have the lines in a hunt group at the telco, but then you're implying you don't care which number was dialed, you just want failover at the telco. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Wednesday, May 23, 2007 8:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FXS + Pots Extensions Help We wanted a cheap last resort fail-over. A few really cheap pots lines are easy to run buy, as we can get them for a really low cost. My understanding with DIDs (and its limited), is they have to belong to a PRI. The only way that is cheaper than a few pots lines is if you needed 8 or more pots lines. Then the line fees balance out. I was hoping for a solution more along the lines of Use this x variable that contains what ZAP channel it came in on, then I can program that one to point to a particular person. Thanks, Rob Sean M. Pappalardo wrote: Rob Schall wrote: Normally I just use pri's with our asterisk systems, but a request came in to add some normal pots lines to the setup. We have 3 lines, and they run into the fxs ports. They hit the dialplan just fine, and they always dial the s extension. However, my question would be... Is there a way to determine what number was dialed and have it forward to a specific phone? Sure, it's called a DID trunk. It's basically just a regular analog phone line but the CO switch sends down the digits dialed in one of a few ways: Dial pulse (DP), Multi-freq (MF), Dual-tone multi-frequency (DTMF). They are usually inbound-only, but some CO's can add outbound service too if needed. Call your phone service provider's business office and ask about analog DID lines/trunks. They should be around $30/mo for the line and $1-2/mo for each number. Ask them what type of signaling they use then you'll need to configure your zapata.conf to match. After that, you can then start routing in the dialplan based on the number called. For extra fun, have the CO set them up in a hunt group to avoid busy signals. Take a look at: http://www.voip-info.org/wiki-Asterisk+tips+DID (BTW, Why are you adding analog lines if you're already big enough for a PRI? Isn't it less expensive to just add a couple more DID numbers to the PRI?) Sean - This E-Mail message has been scanned for viruses and cleared by SmartMail from Smarter Technology, 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- 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
