[asterisk-users] outbound calls on PRI all congested
Hi,
I am just trying to but an Asterisk server between a German Telecom
(Deutsche Telekom) PMUX and a Siemens HiCom. Inbound Calls (Telekom ->
HiCom) work like charm. Outbound calls however all end up being
congested:
-- Executing Dial("Zap/50-1", "Zap/g1/06151343|180|WT") in new
stack
1 -- Making new call for cr 32784
-- Requested transfer capability: 0x00 - SPEECH
1 > Protocol Discriminator: Q.931 (8) len=42
1 > Call Ref: len= 2 (reference 16/0x10) (Originator)
1 > Message type: SETUP (5)
1 > [04 03 80 90 a3]
1 > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
1 > Ext: 1 Trans mode/rate: 64kbps,
circuit-mode (16)
1 > Ext: 1 User information layer 1: A-Law
(35)
1 > [18 03 a9 83 81]
1 > Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive Dchan: 0
1 >ChanSel: Reserved
1 > Ext: 1 Coding: 0 Number Specified Channel
Type: 3
1 > Ext: 1 Channel: 1 ]
1 > [6c 09 00 83 33 36 37 38 33 34 37]
1 > Calling Number (len=11) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
1 > Presentation: Presentation allowed of
network provided number (3) '347' ]
1 > [70 0d 80 30 36 31 35 31 36 36 38 34 33 34 33]
1 > Called Number (len=15) [ Ext: 1 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0) '06151343' ]
1 > [a1]
1 > Sending Complete (len= 1)
-- Called g1/06151343
1 < Protocol Discriminator: Q.931 (8) len=9
1 < Call Ref: len= 2 (reference 16/0x10) (Terminator)
1 < Message type: RELEASE COMPLETE (90)
1 < [08 02 82 ac]
1 < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: Public network serving the local user (2)
1 < Ext: 1 Cause: Requested channel not available
(44), class = Network Congestion (2) ]
1 -- Processing IE 8 (cs0, Cause)
-- Channel 0/1, span 1 got hangup, cause 44
-- Forcing restart of channel 0/1 on span 1 since channel reported
in use
1 > Protocol Discriminator: Q.931 (8) len=13
1 > Call Ref: len= 2 (reference 0/0x0) (Originator)
1 > Message type: RESTART (70)
1 > [18 03 a9 83 81]
1 > Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive Dchan: 0
1 >ChanSel: Reserved
1 > Ext: 1 Coding: 0 Number Specified Channel
Type: 3
1 > Ext: 1 Channel: 1 ]
1 > [79 01 80]
1 > Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated
Channel (0) ]
1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("Zap/50-1", "44") in new stack
May 19 20:02:45 WARNING[5913]: chan_zap.c:8765 zt_pri_error: 2 Cause
code 44 not allowed when disconnecting an active call. Changing to cause
16.
May 19 20:02:45 ERROR[5913]: chan_zap.c:2930 zt_hangup: What is wrong
with you? You cannot use cause 44 number when in state 6!
-- Hungup 'Zap/50-1'
1 < Protocol Discriminator: Q.931 (8) len=13
1 < Call Ref: len= 2 (reference 0/0x0) (Terminator)
1 < Message type: RESTART ACKNOWLEDGE (78)
1 < [18 03 a9 83 81]
1 < Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive Dchan: 0
1
RE: [asterisk-users] bristuff error: "received SETUP message for callthat is not a new call"
On Monday, November 27, 2006 10:23 AM Louis-David Mitterrand wrote: > Hello, > > With the following setup: > > - asterisk 1.2.13, > - zaptel 1.2.10 > - bristuff 0.3.0-PRE-1v > - quadbri card, Have you tried using bristuff 1v with the qozap driver of 1s? All qozap versions after 1s had serious problems (which seem to be fixed in soon to be released 1w). If this does not help, do a pri debug or better yet pri intense debug, describe the problem and contact the author with this info. Regards, JP ___ --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] Snom 360 / firmware 6.5.1 / registration problems with Asterisk
Hi, we just upgraded from 1.2.10 to 1.2.13 and now encounter strange problems with our snom phones (FW 6.2.3 to 6.5.1). Upon phone boot everything works fine. Phone registers and asterisk is happy. Soon afterwards the registration is lost however. Sometimes after a few minutes the phone reregisters, sometimes not. This only seems to happen on the first configured line. Switching back to 1.2.10 solved the problem. What changed between those to versions? Maybe a new setting on the snoms we have to take care of? Funny thing: I set defaultexpiry=60 and told the phone to use 1min as well. After the phone registered I watched the expiry counter with sip show peer. It counted backwards from 60 to about 40. Then it jumped to 70, counted to 0 and the phone was gone. This is somewhat reproducable. And it simply does not look right... Any ideas? Kind regards, JP ___ --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] CDR question with SIP/IAX trunks
Ok... I got it. Someone changed the CDRs to reflect CALLERID(ANI) instead of CALLERID(number) in 1.2.10. According to the release notes this was taken back in 1.2.12. I do not know why this was not done for IAX as well so it would have been consistent at least but well... I am either going to set ANI now or upgrade to 1.2.12... :-) ___ --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] CDR question with SIP/IAX trunks
Sorry for replying to my own post: I just switch the connection from Asterisk A to Asterisk B from SIP to IAX without changing anything else (dialplans on both system are the same). Now the correct callerID is logged. The behaviour changed from 1.2.9 to 1.2.10 I suppose since this worked without problems in August. After my switch to the new version, this started going wrong... Any ideas? ___ --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] CDR question with SIP/IAX trunks
Hi, scenario: Call comes in via ISDN BRI on Asterisk A. Callerid (set by zapata) is let's say 0151123456789. In the incoming context I prepend a 0 to that callerid. My snom correctly displays 00151123456789. The call is also "forwarted" to Asterisk B. On the incoming context of Asterisk B I prepend yet another prefix 98. The callerid now is 9800151123456789 which is correctly displayed on the SNOM on Asterisk B. So far so good. The CDR on Asterisk B logs the callerid 001511234567890 though and ignores the prepended 98. Any ideas why? Kind regards, JP ___ --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] Experience Patton BRI gateways and Asterisk?
Hi, can anybody comment on patton inalp voice gateways and Asterisk? How good is there echo cancellation? How good is the interop with Asterisk? I am especially looking for reports on 4630 and 45xx series with BRI. Thanks a lot in advance! Kind regards, JP ___ --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] Blind transfer 3/4 digits
On Tuesday, September 05, 2006 2:06 AM Ronald Wiplinger wrote: > In my opinion Asterisk remembers all numbers and therefore it does > not wait for the 4, since it found a match. This is in VoIP (in my If both phones enter the dialplan the same way and one phone does work then it should not be a problem with the dialplan or with the way Asterisk is doing the match. You pointed that out yourself. AFAIK there is no overlapping in the dialplan. Either the phone (when dialing, doing a SIP transfer etc.) or Asterisk (when doing an attended/unattended transfer) is waiting the specified time for more digits. If no other number is received it then feeds the received number in the dialplan. So either your phone is just transmitting 601, Asterisk only understands 601 or you do have a problem with your dialplan. The only other option would be a general dialplan bug which is not too likely since most of us would have run into the exact same problem. What do debugs on the Asterisk show you? Do a SIP debug etc. Are you using inband or outband DTMF? ___ --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] Blind transfer 3/4 digits
On Monday, September 04, 2006 3:22 PM Ronald Wiplinger wrote: > What's happen to you guys? Nothing. Why? > I am not yelling, just asking. Maybe in a bit stressed out kind of way. > It is sure not a dialplan question! Without having all necessary information that is hard to say. Maybe one phone comes in a different context than the other etc. Lot's of things that could go wrong in the dialplan. > If it would be a dialplan > question, than it would be for each dialing, but it isn't. If we are talking about the same context and same way of dialing: True. > You mentioned SIP message and that makes me wonder! Are we not using > here dtmf ?? I somehow had the impression that you are using the transfer button on the SNOM which would tell the SNOM to transfer the call. You are obviously talking about attended/unattended transfer via Asterisk only, correct? Then ignore my suggestion. > If it is a sequence of "tones", Well... If you are using inband DTMF: correct. Otherwise DTMF may correspond to SIP messages as well but let's not get into that. I suppose you are using inband DTMF and G.711? > than why is it different if it is in > a string (like snom) or another phone, with single tones? If the dialplan is not responsible obviously the phones are behaving differently. Maybe the DTMF sequence is not transmitted correctly but on the other hand I am using SNOMs with inband DTMF without any problems. Maybe the phone (as others suggested) is doing some number/pattern matching magic which you have to fiddle with. > If we understand this part, than is the question, where can I turn on > the system to take a longer break between "tones" still as a string? The default setup should not be a problem with SNOMs (at least I never read anything about it) but have a look at the features.conf options. > That should proof my thoughts (and that > without yelling, ... hehehehe) But a lot of exclamationsmarks. :-) Just kidding. As others pointed out: We (at least I) would need the entire picture (the relevant parts of your dialplan etc.) to really help you here. Regards, JP ___ --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] Blind transfer 3/4 digits
On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote: > try that way. However, I have doubts as well. If you are right, than > why snom phone does not have this problem? Would not here also the > first match count? Because the transfer button on the SNOM is using a totally different mechanism than sending # to Asterisk. On your snom configuration (like ours) the phone does not start to create/send a SIP message until you hit "OK". At that time the entire number is there and a complete SIP transfer is created. Cool down a bit. The problem you are having is most probably just a dialplan problem. It takes some time and experience to get those things right. No need to yell here... ___ --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] transfer call von D-channel
On Thursday, August 10, 2006 10:43 AM Christophorus Laube wrote: > Hi list, > > how can I realize explicit call transfer? I want to transfer a call > which I answered to another phone and it the other one answers I want > to hang up so that my resources are freed. Is that possible with > Zaptel or which channel can I use else? Should work with current bristuff and Zap if you enable ECT in zapata.conf. I am having some problems with Siemens Gigaset but it should work. ___ --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] Echo cancell
Switch your echo canceller to MG2. Look for zconfig.h in zaptel and recompile. Also experiment with echocanel=256 and adjust rxgain/txgain. ___ --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] BRIDGEPEER and DIALEDPEERNAME empty
Hi, an all of my installations (1.2.9.1 bristuffed) the parameters BRIDGEPEER and DIALEDPEERNAME are empty after a successful dial command. Can someone please try to confirm this? I am not sure what I could have done to the implementation to cause this so it might very well be a bug. Thanks, JP ___ --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] cmd DIAL - Who picked up the call?
On Wednesday, August 02, 2006 8:35 PM Vadim Berezniker wrote: > No idea, but DIALEDPEERNAME should contain the same value as > BRIDGEPEER. Try that. Already did. Both are empty. ___ --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] cmd DIAL - Who picked up the call?
On Wednesday, August 02, 2006 7:39 PM Vadim Berezniker wrote: > DIALEDPEERNUMBER contains the exact peer spec for the peer that > picked up. You can use that. Consider yourself my hero of the day! That looks VERY promising. It does not show the technology so Dial(SIP/phone_200&Zap/g2/13,,M(getchannel)) will return either phone_200 or g2/13 but I will find a solution for that as well I suppose!!! Any idea why BRIDGEPEER is empty here all the time? Kind regards, JP ___ --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] cmd DIAL - Who picked up the call?
On Wednesday, August 02, 2006 6:49 AM Eric "ManxPower" Wieling wrote: > Zap/10-43 would indicate that this is the 43rd call (call waiting) on > channel 10. Obviously this would have to be removed to do it the way > you want. Obviously. :-) Or we find another solution for the problem/challange... Ideas? ___ --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] cmd DIAL - Who picked up the call?
On Tuesday, August 01, 2006 9:36 AM Kai Ober wrote: > when you park a call (asterisk feature defautl keys: #700 ...) at > your isdn phone and you "forgot" to catch the call on another phone, > the phone from where you parked the call, should ring after 45 > seconds (default) > does this work for you? (which asterisk version dou you have?) 1.2.9.1 bristuffed and no it does not seem to work. It seems to mixup src and dst channel: == Parked Zap/4-1 on 701. Will timeout back to extension [from_internalisdn] s, 1 in 300 seconds The call came from another extension and another context. Therefore the callback will fail (and _does_ fail)... Will you file a bug report and give me the bug number? ___ --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] cmd DIAL - Who picked up the call?
On Montag, 31. Juli 2006 6:27 William Piper wrote: > Doesn't the dstchannel in the CDR's show this already? Does not work for zapata BRI/PRI combinations. If I call Zap/g1/43 e.g. and group 1 is span 1 and 2, dstchannel will be something like Zap/1-1 or Zap/2-1. Imagine Dial(Zap/g1/43&Zap/g1/44) A simple Zap/1-1 in dstchannel will tell you nothing... We are using dstchannel for IAX/SIP calls though so you suggestion is of course valid. With the exception of zapata and ISDN. Kind regards, JP ___ --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] cmd DIAL - Who picked up the call?
On Monday, July 31, 2006 3:12 PM Kai Ober wrote:
> (How do you get to the dial command, can you send the extension for
> this?)
>
> the idea is to to use $EXTEN.call a macro with $EXETN as an argument
> ...
The problem is this:
exten => 43,1,Dial(SIP/phone_1&Zap/g1/43)
I need to find out who picked up the call not who was originally called.
${EXTEN} will always contain 43 in this scenario. It will not help you in
pickup or delayed dial scenarios in which phone A rings 10 seconds and after
that phone B and phone C start ringing as well. We need to see if the customer
acutally spoke to user A, B or C regardless of who was called originally.
Kind regards,
JP
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RE: [asterisk-users] Multiple dialing
On Monday, July 31, 2006 9:22 AM Marcus Carlson wrote: > So, my question is; How can I make SIP/ext1 call continual all the > time? Use a local channel. Have a look at voip-wiki cmd Dial. There should be an example there. Basically you create a local channel in your dialplan that first waits for x seconds and then dials the second extension. Then you dial you first extension and the local channel simultaneously. Regards, JP ___ --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] SNOM 360
On Friday, July 28, 2006 3:08 PM Dovid Bender wrote: > I am trying to have thier PC run thru the port on the phone and the > phone give prioroty to itself and the rest to the PC. When my client > does a big download the phone call gets real bad. The docs from SNOM > on TOS (or DIFFSERV) is poor and I dont understand it well enough. > Anyone have configs or docs on how they did this ? I would be surprised to learn that the Snom is actively doing traffic management itself. Traffic managment must be done at the bottleneck to be halfway successful. Let's assume you are doing a download and you snom would do traffic management giving itself priority. What if your co-worker is doing a huge download? How should your snom know and throttle his download? No way. ___ --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] cmd DIAL - Who picked up the call?
On Friday, July 28, 2006 3:12 PM Kai Ober wrote:
> What about DIAL ( |M(macro-name))
> and set the userfield in cdr during execution, ...
Set the userfield to what? That is the entire problem. ${CHANNEL} will give me
something like Zap/10-1. ${BRIDGEPEER} is empty. I would love to see the called
MSN in the port-field something like Zap/10-43 if MSN 43 was called... :-) That
would help enourmously.
Kind regards,
JP
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[asterisk-users] cmd DIAL - Who picked up the call?
Hi, if I do Dial(SIP/peer1/number&Zap/g1/Number) can I somehow figure out who exactly picked up the call? In the cdrs dstchannel I can see the channel but not the extension dialed. E.G. a Dial(Zap/10/43) will result in a CDR Zap/10-1 which does not help me unfortunatly. Any ideas? Kind regards, JP ___ --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] SNOM 360
On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote: > Does anyone know how to set up QoS on the SNOM 360 ? Thanks. What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a Snom 360 that will manage things for you. AFAIK all you can do is tell the phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the task of the equipment managing the bottleneck (firewall, router whatever) to use this information and manage your traffic accordingly. Regards, JP ___ --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] CDR dest question
Hi, we are having some trouble with CDR records. Example: Case 1: Customer 12345 calls extension 10. Extension 20 takes the call using Pickup (e.g. *810). I now have two CDRs: 1: 12345 -> 10 2: 20 -> *810 I could live with the second CDR but the first gives the impression as if 12345 was talking to 10 while in real life he/she was talking to 20. How can I fix this? Case 2: We are using Dial commands with several channels e.g. Dial(Zap/10&Zap/20). I cannot see what channel (and therefore the associated default extension with that channel) picked up the call. Is there a way to fix this? Kind regards, JP ___ --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] How many users on an asterisk box behind a dsl canyou have
On Montag, 17. Juli 2006 6:40 ted jones wrote: > I have been trying to read up and understand Asterisk. I have a > small office of 25 people growing to 50 and have a dedicated DSL for > Asterisk What kind of DSL? Synchronous, Async? What speed? > and another DSL for computer use and was wondering using gsm > primarily how many users I could put on the asterisk box on a single > dsl. What kind of box? E.g. 50 concurrent calls on a small VIA might be a problem. Do you plan to have the calls go through Asterisk or use it only to connect the SIP endpoints? Are you planning on monitoring/recording calls? > Average calls is probably going to be 25-35 at any given time. You are talking about concurrent calls? ___ --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] Snom 300 headset with static noise
On Freitag, 14. Juli 2006 10:13 Adrià Vidal wrote: > Someone using these phone Snom 300 with his own headset ? We used to but the quality was horrifying. Since we changed to Plantronics Noise Cancelling headsets everything is wounderful. > We got horrible static noise on them? Maybe the article Michiel pointed out helps you still the overall voice quality of their headsets (at least the ones they sold last year) is awful. Kind regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Out of Office Auto Reply:
On Friday, June 23, 2006 4:08 PM Steven wrote: > Exchange changes > > http://www.microsoft.com/exchange/techinfo/tips/mailtip01.asp Looks promising and helps a bit. Still no use of precedence bulk etc. though. Very poor detection of "lit" mails. ___ --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] Out of Office Auto Reply:
On Thursday, June 22, 2006 8:13 PM Anthony Rodgers wrote: > We use MS Exchange too and, as far as I am aware, it is cognizant of > mailing list headers and doesn't send OOO notices to mailing list > postings. The only mailing list from which I receive my own OOO > notices is one that doesn't have the proper mailing list headers set. No. Exchange does not honour "Precedence" headers. It has some funky way of determining what is a mailing list and what is not. It does not work very well and it has (or had) to be enabled via a registry key. If you don't do this, even Exchange 2003 will reply to some mailing lists. But it should not send this to every mail but only once day... Kind regards, JP ___ --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] Snom 360 with Firmware 6.1?
Hi, Has anybody experience with Snom360 and Firmware 6.X with Asterisk 1.2.X? I am currently using Firmware 5.5 without serious problems but wanted to make sure 6.X will work as well (including subscription etc.) Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] skype out
On Tuesday, June 06, 2006 12:10 AM Andrei (MPI) wrote: > I'm using SIP-to-Skype/Skype-To-SIP software gateway called Uplink > (found in Wiki): http://nch.com.au/skypetosip/ - which is free and > working great so far. Downsides are: Only that it produced not RFC conform SIP headers which are blocked by some firewalls (e.g. Juniper). I tried to contact their technical support several times but have not received any answer whatsoever. Very poor support Kind regards, JP ___ --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] Bristuffed Asterisk: Hangup problems
On Friday, May 12, 2006 12:38 PM stoffell wrote: > I have sent an email to junghanns.net about this, but haven't > received an answer yet. If I do receive anything, I'll post it back > to the thread. Friendly piece of advice: Call them! :-) Mail is most of the times not answered... Regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk hardware
On Sunday, May 07, 2006 10:07 AM Steve Totaro wrote: > Oooops, sorry its late. Obviously. :-) > My favorites in order, Polycom, Snom, Cisco. Since when do these use IAX? He asked for IAX hardphones... If I am mistaken let me know since I am looking for good reliable SNOM-like IAX phones as well! :-) Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Transfer - context/priority
On Wednesday, May 03, 2006 8:56 AM Tomislav Parcina wrote: > I have find answer. It transfer's the call to context defined in > sip.conf file. Now, I have another question, is it possible to define > some other context? If it isn't than this could be nice feature > sip.conf trcontext=sip-transfer You can do Set(TRANSFER_CONTEXT=transfer_ctx) In the dialplan. Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, sodisappointing !)
On Saturday, April 15, 2006 3:17 PM Remco Barende wrote: > I heard that Junghanns is working on such an interconnection. It is > already possible to connect their PRI cards, and they are working on > BRI<->PRI. Correct. The next driver generation is supposed to support this fully. > I ise their bristuff for an HFC-S BRI card and am not happy at all > with the way they implemented timing, without applying the florz > patch I have lots of problems (lockups, lost line etc.) > > My hesitation is with the driver, I think florz only fixes HFC-S if I > would run into similar trouble with PRI I would be in deep trouble. The drivers for their BRI/PRI cards are totally independant from the standard HFC driver. I own a QuadBRI card and do not have any timing problems whatsoever. > But it would certainly fix faxing. Bristuff will help anyway. You can disable echo can and modulations (rx/txgain) on a per call basis which is very important for faxing and bridged call faxing. Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] question about DISA
On Sonntag, 9. April 2006 9:58 Tele Cost Price Reducer wrote: > hi Ronald, > > i would use a CallerIDNum authentication, based on the Asterisk > Database to solve it. > > then you do not need any verification. Dangerous. CallerIDs can be easily faked in some countries using VoIP providers. Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Junghanns and Digium TDM400?
Simple: Do not use E-Mail. Phone him or try Instant Messaging. kapejod does not really read mail... :-) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier KriefSent: Tuesday, April 04, 2006 8:50 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Re: Junghanns and Digium TDM400? 2006/4/3, Koopmann, Jan-Peter <[EMAIL PROTECTED]>: Have you contacted Junghanns directly? They do not read this list. Usuallythey are very helpful in such issues. Phone them.What is your secret to get a reply from Junghanns ? smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma
On Monday, April 03, 2006 5:51 PM Andrew Kirch wrote: > I've never had issue with the Digium cards in testing and as we're > looking forward to production systems what compelling reason do I > have to pick Sangoma? (I'm not looking for a flame-fest here, but Never used either card myself but Sangoma is supposed to have the by far better echo can hardware. 128tap per channel and not per card/span/whatever. Look in the archives. Many people claim they had echo problems they were not able to solve with Digium hardware that simply went away the moment they used Sangoma. Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.
On Friday, March 31, 2006 3:52 PM Chris Earle wrote: > I'm wondering if I should be using zapHFC with my Junghanns card > instead of qozap? Why would you want to do that? Sorry if I missed the start of the problem but qozap is what you want to use with your Junghanns card (at least if you want to stay with zaptel/libpri). > Everyone always mentions zaphfc -- mostly I > guessed because they are using a zaphfc-compatible card - but > *maybe* I should try that instead of qozap??? No. I would use zaphfc only for cheap 1 BRI HFC cards. All Quadro or Octo BRI cards I know of use their own drivers and need them! Note that the HFC chipsets differ on those cards and in order to use the features of the more complex HFC chipset on Quadro/OctoBRIs I assume you will have to stick with qozap. Especially when it comes to bridging calls between BRI interfaces or even cards (I think this will be part of one of the next driver versions from Junghanns). Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Junghanns and Digium TDM400?
On Thursday, March 30, 2006 8:27 PM Chris Earle wrote: > Someone plase help > This is terrible. Been trying everything for weeks :-( Have you contacted Junghanns directly? They do not read this list. Usually they are very helpful in such issues. Phone them. Mit freundlichen Grüßen Jan-Peter Koopmann Dipl.-Wirtschaftsinformatiker Geschäftsführer -- Seceidos GmbH&Co. KG | Tel: +49 6151 66843-43 Robert-Bosch-Str. 7| Fax: +49 6151 66843-52 64293 Darmstadt / Germany | IAX: [EMAIL PROTECTED]/43 http://www.seceidos.de | SIP: [EMAIL PROTECTED] smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speed up dial using #?
Hi, since we do not have a nice common numbering plan like (XXX) XXX for national phone numbers here in Germany, the dialplans usually contain lines like this exten => _0X.,1,NoOp(Dial outwards etc.) If you use such context with overlap dial (DISA, ZAP), it takes a while for Asterisk to recognize that the number dialed is actually complete and can be processed. I understand why this is the case and this is a common problem not only for Asterisk. AVM folks are used to simply add a "#" after the last digit to let the Fritz!Box know that the number is now complete and the Fritzbox starts to dial the number immediatly. Is there a way to do something similar in Asterisk? Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Native MOH - Convert mp3 to ulaw
On Tuesday, March 21, 2006 9:36 PM Douglas Garstang wrote: > I tried that earlier today... found it somewhere online... This is > what I get... > > [EMAIL PROTECTED] mp3]# sox -V fpm-calm-river.mp3 -t au -r 8000 -U -b > -c 1 fpm-calm-river.ulaw resample -ql sox: resample opts: Kaiser > window, cutoff 0.94, beta 16.00 > > sox: Failed reading fpm-calm-river.mp3: Do not understand format > type: mp3 Not all SOX versions support mp3. Some distros (at least Suse) compile sox without mp3 support. Either compile your own version with mp3 or convert the files to wav first. Regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR problem with TAPI
Hi, we just noticed a strange CDR problem. We are using individual phone numbers for all our SIP phones. During dialout we do a database lookup in order to set the correct callerid (e.g. phone has number 100 but in external calls this should be displayed as CID -20). This works like a charm and the CDRs look correct. When we start a call using TAPI (e.g. AstTapi) the call setup is a bit different: I start the call, my phone (100) rings, I fetch the call and Asterisk dials the intended number. In the LOG I can see that the callerID is again set correctly to 20. But the CDR now does not show from 20 to 1234567 but from 100 to 1234567 ignoring the Set(CALLERID(number)=20) completly. Bug? Feature? Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Echo Cancellation
On Wednesday, March 15, 2006 9:49 PM Hagen Rode wrote: > I've got exactly the same problem with echo, where the mic feeds into > the speakers. I'm looking at purchasing the Tellabs 2572 64ms T1 echo > cancellation card to see if it will help. Are we still talking about people attending MeetMe conferences with laptops but without headsets? If so: Personally I doubt that a 64ms echo can will help you with this issue. I have not yet measured this but am willing to bet that the echo produced by such a speaker/mic setup will exceed 64 and even 128ms. Perhaps you might want to record a test and have a look at the WAV file to at least get an idea of how long the echo really is before you invest in any sort of echo can hardware. You might very well find out that your only option are headets. Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic
On Thursday, March 09, 2006 8:18 AM Douglas Garstang wrote: > By 'code for asterisk' are you referring to the Asterisk source code? > If so, step back and think about your statement for a moment. If, for > Asterisk to be enterprise class, it's source code needs to be > modified from it's current content, it's hardly enterprise class, is > it? What is this thread all about? Is Asterisk "enterprise class"? The answer is obvious: It depends on your definition of "enterprise class". If you definition includes things like "RTP in/out traffic on multiple interfaces must work" then the answer is no. If you definition is somewhere along the line "can be used in most enterprises without problems" then the answer is yes. If you need a feature you at least have the possibility to code it yourself (yes, source code). Avaya&Co give you the opportunity to hand in a feature request but nothing more. Unless you pay for the feature they will probably not implement it and you have no way of doing so yourself. Asterisk does not really meet my personal definition of "enterprise class" but since there is no commonly accepted definition in the first place, why trust Digiums words on the website at all? I usually do not trust any marketing phrase like that no matter what the product is. Try Asterisk yourself and if your decision is that you cannot use it, then don't! But please stop getting on peoples nerves bashing on the term "enterprise class". It will not get you or us anywhere. If you are not satisfied with what Asterisk can achieve you have plenty of other choices. Feel free to use them. Feel free to contribute to the project. Constructive criticism is wanted (at least in most of the cases this seems to be true even though there is room for improvement, agreed). Currently you are not helping at all. Annoying is a term that comes to mind though... Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: MOH native files
On Thursday, March 02, 2006 11:47 AM Tomislav Parcina wrote: > sox: Failed reading fpm-calm-river.mp3: Do not understand format > type: mp3 > > Have I done anything wrong? Well your sox does not understand mp3 since the support is not compiled in. Compile your own suitable version of sox. Regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk transfer conflict
On Wednesday, March 01, 2006 3:52 PM Fredrik Jensen wrote: > " # " is also used as a transfer button on my asterisk, so when I > press it I hear my Asterisk trying to transfer the call. Have a look at features.conf. What version of Asterisk are you running? Staring with 1.2 (I think) you can map transfer to anything you like (e.g. ## instead of #). Regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to add stun functionality in asterisk
On Friday, February 17, 2006 7:34 PM Matt wrote: > Yes Sir! This is what I use: > http://www.vovida.org/applications/downloads/stun/ > > Works like a charm! Been running it in production for about a year. Good hint. Can you possibly provide a bit more insight on this? Are you running STUN so that your phones behind NAT can easily connect to your server or the other way around? I would really like to see the relevant parts of your setup. Kind regards, JP ___ --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] SIP and firewalls?
On Wednesday, February 15, 2006 1:59 PM John Jensen wrote: > Hi Hagen, > It's not exactly a pleasure to run SIP through firewalls but it can > be done. > At least in under some circumstances. If you use a decent Firewall it will analyze and interpret the SIP Headers etc. and open the correct ports for you. Only NAT trouble left then. Again... A decent Firewall will help! Regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Snom 360
On Wednesday, February 15, 2006 12:42 PM Garth van Sittert wrote: > Could we possibly see your settings to get this right? I am trying > to get it working at the moment. > I can see the phone buttons have subscribed to asterisk, but they > just don't light up. We are using 4.1 firmware and are upgrading to > 5.3 to see if it helps. No problem here with bristuff and SNOM (both FW 4.x and 5.x). Have you set your hints correctly? Regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] attended call transfer
On Sunday, February 12, 2006 9:36 PM John Novack wrote: > That certainly is the way it SHOULD work. Blind and attended transfer > should be able to be initiated the same way. ... > I would consider that a defect or bug, not a new feature request. I second that. Regarding the bounty: Once someone can at least make an educated guess as to how much this improvement would cost and if there is some sort of guarantee that this functionality will make it to Asterisk soon and not longer in SVN for several months, I am sure there are enough people kicking in a few bucks to make this happen. Kind regards, JP ___ --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] Dynamically disabling echo cancellation (Zap).
> Hi! For reasons that I won't bore people with, I'd like to disable > echo cancellation on-the-fly, depending on which DID a call came in > on. I've seen things like spandsp disable EC for faxes, so I know > it's possible. Any idea where to start looking? (I assume I'll have > to make a helper application of some sort to be called externally, > and that's fine.) If you are using bristuff you could use the m option Dial(Zap/g1m/Number) in order to get rid of rxgain/txgain and AFAIK echo cancellation. This has been built into bristuff esp. for faxes etc. Regards, JP ___ --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] trunk to trunk forwarding
Would that not be ECT (Explicit/External Call Transfer) for BRI/PRI? If your telco supports this (which many do), all you need is ECT support for your driver. I think bristuff ZAP is either able to do it now or will be soon. Regards, JP ___ --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] MeetMe Dialplan question
On Saturday, January 21, 2006 10:35 PM Philipp von Klitzing wrote: > Look at this: > http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+voicemail+liv e > > That is very close to what you want to achieve, maybe even a little > overkill, but I think it'll get you there with a little tweaking. Wooow. Looks interesting. But I am afraid I will need a few more coffees before I tackle that one! :-) > Your other option would be to e.g. use one of the SNOM buttons to > call an action URL (in the middle of the call) that uses the manager > API to transfer/bridge certain channels. Or simply play with the > codecs of your SNOM and find a codec where you don't get distortion > with the phone-based 3pty conference. Looks like the way to go. I hope I will find time to wake SNOM regarding this support request. :-) Regards, JP ___ --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: Re[2]: [Asterisk-Users] Re: MeetMe Dialplan question
On Saturday, January 21, 2006 8:02 PM Alexander Chemeris wrote: > What is the problem with step 3? > > See this example as basis for modifications: > http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro Unless I have terribly misunderstood that macro, that is basically the same thing I am doing now, is it not? Simply transfer the customer to a conference room (I might have a look into the automatically determined conf room number), then transfer all collegues in there as well and finally jump in myself. It is however not quite what I described in step 3. Thanks, JP ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: MeetMe Dialplan question
> I think the solution needs a little more thinking about I am reliefed. I almost thought I had missed something that obvious... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: MeetMe Dialplan question
On Saturday, January 21, 2006 1:44 PM Tony Mountifield wrote: > - edit the code to provide another option to turn off that message, or http://bugs.digium.com/view.php?id=6316 Regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: MeetMe Dialplan question
Hi Tony, > Look at the option 'G(context^exten^pri)' in the Dial application. Thanks for the hint but I am not sure if this will help me. Either I am too blind to see the solution or I stated the question in an unclear way. :-) What I want is this: 1. Customer calls me or I call customer. 2. In the middle of the call I decide to get an additional collegues in the call. Usually I would put the first call on hold, call the collegue and then press the conference button on my SNOM 360. Unfortunatly there seems to be a problem with the SNOMs and Asterisk 1.2.x since audio on those conferences get distorted after a few seconds. Therefore I need a substitution for this using MeetMe. I thought about this: 3. I transfer the call to my "personal" MeetMe room. In this step I would like not only the customer but also me to be connected to the MeetMe room automatically. Basically I can continue to chat with the customer without him noticing anything. 4. I now put the call on hold and call the collegue. If he wants to join I simply transfer him to the room as well and can continue to do so with other collegues. In order to return to the conference myself I now do not need to call the conference number myself but simply return to the call created in step 3. With the exception of step 3 everything seems easy. How can I solve this with the G-option? > Specifying the 'q' flag to MeetMe disables it. However, it also > disables all the enter/exit sounds and so on, so if you still want > those you will have to either: Yep found that but as you said it disables all sounds. > - edit the code to provide another option to turn off that message, or Can't be too hard. I will have a look at the code an provide a patch. > - replace the sound file conf-onlyperson.gsm with a file of zero > duration (not necessarily just an empty file - it might still need > a header). Since I want this for usual conferences this is not an option I guess. :-) Kind regards, JP ___ --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] MeetMe Dialplan question
Hi, is the following possible? I would like to transfer a call to my "personal" MeetMe conference room and get transferred there automatically as well. Currently I can transfer the call to the conference, have to hangup and then call the conference number myself. I would love to have this in one quick function. Moreover is there a way to disable the "You are currently the only person in this conference" prompt for the first user? I know how to enable/disable this for the following users but not for the first user. Thanks in advance, JP ___ --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] Disabling zap echo cancellor from dialplan
On Thursday, January 19, 2006 6:53 PM Massimo De Nadal wrote: > Anybody knows if it's possible to disable zap echo cancellor from > dialplan only for certain outbound calls ?? > > I share the same phone lines for voice calls and faxes. Iaxmodem > works fine for me only turning off the echo cancellor, but I need it > for voice calls. Any ideas ? If you are using bristuff add the m option like this: Dial(Zap/g1m/numbertodial) This will turn off the modulation (rxgain/txgain) and should (!) turn of the echo cancellation but I am not sure about this. Regards, JP ___ --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] RX/TXgain on bristuff/zaptel ?
On Tuesday, January 17, 2006 4:13 PM Pisac wrote:
> You are right, only outgoing calls!
> I found lines that you mentioned, but I do not understand where is
> difference? In current chan_zap.c I read:
>
> if (!IS_DIGITAL(ast->transfercapability)) {
> set_actual_gain(p->subs[SUB_REAL].zfd, 0, p->rxgain, p->txgain,
> p->law); } else { set_actual_gain(p->subs[SUB_REAL].zfd, 0, 0, 0,
> p->law); }
>
> And your suggestion is:
>
> if (IS_DIGITAL(ast->transfercapability)) {
> set_actual_gain(p->subs[SUB_REAL].zfd, 0, 0, 0, p->law); } else {
> set_actual_gain(p->subs[SUB_REAL].zfd, 0, p->rxgain, p->txgain,
> p->law); }
>
> Which is the same thing but invertedly written. I'm not a programmer,
> so I may be wrong (maybe IS_DIGITAL could be NULL), but I would like
> to understand difference in those two segments.
I know. It sounds crazy. Trust me though. Somehow the original does not work.
This fix is from the bristuff developer himself and it does work.
Regards,
JP
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[Asterisk-Users] How to find out if a new voicemail exists
Hi, I would like to see if during a call a new voicemail was recorded. I want to send a SMS to mobile phones if someone recorded a message on our voicemail system. I can use VMCOUNT to see if there are new messages in the Inbox but this will result in new SMS being sent even if the caller hangs up during the Voicemailpromt, at least if there are still unread/unheard messages in the inbox. Is there some option or variable I missed or is this a feature request? Kind regards, JP ___ --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] Pickup Button
On Monday, January 16, 2006 11:32 PM C F wrote: > Use callgroups, pickupgroups, and a programmed speeddial programmed > to the pickupexten in features.conf for the button. Or use Pickup() (or Dpickup if you are using bristuff): http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup Then you will not need callgroups. Regards, JP ___ --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] RX/TXgain on bristuff/zaptel ?
On Sunday, January 15, 2006 3:41 PM Pisac wrote:
> Do bristuffed zaptel (zaphfc) supporting rxgain/txgain in zapata.conf?
Please recheck: This happens on outgoing calls only, correct? On inbound
calls rxgain/txgain work? There is a bug in current bristuff: Somwhere
around line 1928 you should find a few lines regarding gain. Change them
to look like this:
if (IS_DIGITAL(ast->transfercapability)) {
set_actual_gain(p->subs[SUB_REAL].zfd, 0, 0, 0, p->law);
} else {
set_actual_gain(p->subs[SUB_REAL].zfd, 0, p->rxgain,
p->txgain, p->law);
}
Recompile and things will start working on outgoing calls as well.
Kind regards,
JP
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RE: [Asterisk-Users] IAX voice distortion with full upload channel /SIP ok
On Saturday, January 14, 2006 2:45 PM Rich Adamson wrote: > The iax problems tend to be oriented around version issues. Many of > the itsp's have added whatever functionality they needed to asterisk > to support their operation, and upgrading their code to the latest > levels is not a trevial task. Misunderstanding: I am talking about my private * against our company *. Both are running the exact same version of Asterisk. > Given the changes that have occurred in the iax code over the last > year or so, mismatches in iax versions are known to cause significant > audio quality issues. Turning off the jitterbuffer, trunk=no, etc, is > oftentimes the only way to get close to reasonable audio quality. I already tried this and this is not helping. Regards, JP ___ --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] IAX voice distortion with full upload channel /SIP ok
On Samstag, 14. Januar 2006 1:47 tim panton wrote: > That is weird, you would expect IAX to do better than SIP (bandwidth > wise) My point exactly. > 1) are you sure IAX trunking is actually happening ? It shows (T) in iax2 show so I am pretty sure. Timestamps are enabled as well. > 2) what codecs are you using. Are the codecs the same for IAX as > for sip? G.711 alaw and yes the same for IAX and SIP. > 3) is it possible that some of the network hardware is 'sip aware' I strongly doubt it. Our firewall is but only regarding to opening the correct RTP ports for a SIP call. No traffic shaping is done on that end. > 4) How many simultaneous calls are you running between the 2 > endpoints? Happens with one call. > 5) What happens if you turn trunking off ? No change. Kind regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: 1.2.1 "Silence suppression is disabled" whatthehell?
On Sunday, January 15, 2006 12:21 AM Tony Mountifield wrote: > In article <[EMAIL PROTECTED]>, > Pisac <[EMAIL PROTECTED]> wrote: >> I've found something here: http://bugs.digium.com/view.php?id=5374 >> >> but I don't understand how this can be connected to my problem :-( > > It looks like the maintainer of the BRIstuff distribution might have > decided that patch was worth including, even though it is not in the > standard 1.2.1. That does give scope for confusion though! Look at the CHANGES. I was the one who convinced kapjeod to put that patch in the current bristuff distribution. So yes: It is in bristuff as of 1F: 0.3.0-PRE-1f - THIS IS GETTING CLOSER TO A STABLE RELEASE, USE IN PRODUCTION AT YOUR OWN RISK! - merged patch for bug 5697 (meetme) - merged patch for bug 5374 (asynchronous generation of outgoing frames) - _finally_ fixed "sending-nonRFCcompliant-SIP-NOTIFYs" bug (asterisk, extension states) - some debug output clean ups in libpri Kind regards, JP ___ --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] Advice Of Charge (AOC) ?
On Sunday, January 15, 2006 10:32 AM Armin Schindler wrote: > It is not implemented in chan_capi yet, but this is very easy. > The question is what should be done with the AOC information? > Just set some variable? As far as I know Asterisk has no > API/structure for that. At least none that would help you. See http://bugs.digium.com/view.php?id=6152 and put your requests there please. There are a few things that need to be done: 1. Ensure that AOC values survive channel changes. 2. Ensure that AOC information can be retrieved/saved in the dialplan/cdr. Regards, JP ___ --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] rxgain/txgain test numbers in Germany?
On Saturday, January 14, 2006 3:36 PM trixter aka Bret McDanel wrote: > and some employees know what you are talking about and others (most?) > dont. The brightest people are usually working on problems so who > does that leave to answer the phones? Actually I have not called them yet and I seriously doubt I will find someone who remotely knows what I am talking about. :-) I might give it a try next week. Just thought someone else already figured this out and would share the information. Kind regards, JP ___ --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] rxgain/txgain test numbers in Germany?
Hi, does anyone have test numbers in Germany that would allow me to tune my rxgain/txgain settings? I know there are numbers provided by other providers in UK e.g. but have yet failed to find a number in Germany (esp. by Deutsche Telekom). Kind regards, JP ___ --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] IAX voice distortion with full upload channel / SIP ok
Hi, this is the scenario: One * is placed in a central location with more than enough up/down bandwidth. One * is placed behind a DSL 3000/384. Both * are linked via IAX trunking. Everything is fine until the upload channel of the remote site is filled with a download, then heavy voice distortion starts. Well of course this is expected. So I fooled around with HFSC QoS scheduling on the remote site Linux machine. The scheduling seems to work as the TOS marked traffic is put in the correct queue and the upload bandwith for other applications is going down. BUT: The voice quality problems definatly stay when using IAX. The funny part: Doing the same with SIP shows no big problems. SIP calls to T-Online work nicely and even if I change the * <-> * link from IAX to SIP everything is fine even with full up-/downloads on the remote DSL connection. My conclusion would be that this depends on the IAX implementation somehow. I tried different settings for jitterbuffer, trunk and trunktimestamp all with the same result. This currently means we go back to SIP for *<->* linkage. Any ideas? If I should rather post this over in -dev please let me know! Kind regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: AoC (Advice of Charge)
On Friday, January 13, 2006 11:50 AM Zoa wrote: > Is this on bugs.digium.com ? Sure: http://bugs2.digium.com/view.php?id=6152 The bug we chatted about yesterday. Kind regards, JP ___ --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
