Re: [asterisk-users] disable comfort noise
On Fri, 2010-01-29 at 07:29 -0600, Kevin P. Fleming wrote: Szasz Szabolcs wrote: How can I disable comfort noise on Asterisk? Asterisk does not have a comfort noise generator, so there is nothing to disable. You'll have to be more specific about what you are trying to accomplish. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org I expect he means this: rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Something else that is flaky, missing or otherwise irritatingly broken in the piece of shit that is 'Asterisk'. -- _ -- 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] Unable to create channel of type 'DAHDI'(cause 0- Unknown)
On Thu, 2010-01-28 at 23:11 -0600, Karl Fife wrote: Appears completely resolved! No more home-spun patches! Thanks! -K It's *not* fixed here: DAHDI Version: 2.2.1 Echo Canceller: MG2 But as is depressingly the 'norm' for Asterisk it comes back to bitching about hardware 'buy an expensive Digium echo machine instead of a cheap one' rather than the fact that the core of Asterisk is rotten, buggy and the fix usually comes in the form of a developer arguing that it's somebody else's issue. Really - if Asterisk is 'The future of telephony' I can only assume that statement comes from the late 1800's. If you like echo, flaky connections, intermittent service and partially working DTMF coupled with a hefty hardware price tag then hey ho - Asterisk is your man Nice try, be great when it's finished. -- _ -- 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] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, 2010-01-13 at 11:13 -0500, Kristian Kielhofner wrote: On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: ..snip.. I've not been able to get that out of them, but I don't *think* It's Asterisk based because they say: Unfortunately, our assistance with Asterisk is extremely limited. For configuration problems you will have to rely on other sources. [http://www.sipgate.co.uk/faq/index.php?do=displayArticlearticle=540qw=asterisk] Just because they don't offer assistance with Asterisk doesn't mean they don't use it themselves. If you send me a packet capture in PCAP format with SIP+RTP between your system and your carrier I can debug this further. Unfortunately I can't do that - but looking at the captures I can see some slight differences between working and non working scenarios: http://fotobytes.co.uk/user22171/dtmf_debug.php Perhaps you can suggest where I should next look to trouble shoot this? You can try the rfc2833compensate option... Other than that I can't know until I see a packet capture. Tried, but this was not successful :-( What I've done is detailed here: http://fotobytes.co.uk/user22171/dtmf_debug.php Once I get to the bottom of it, I'll write it up properly - for the benefit of other Sipgate/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
Re: [asterisk-users] Sipgate DTMF not detected
On Tue, 2010-01-19 at 13:15 +0100, joern wrote:
listu...@spamomania.co.uk wrote:
I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to
recognize digits pressed on a keypad coming in from a Sipgate trunk.
There answer was to set this:
dtmfmode=rfc2833
in the general section of sip.conf
This has made no difference. I've tried a range of settings (auto,
rfc2833,info) but no matter what, it plain refuses to pick up key
presses.
Locally, if I call from an extension on an ata or a softphone, it works
flawlessly (I have no fxo, everything is SIP based).
It's extremely frustrating and I would be grateful if anyone could offer
some help troubleshooting and fixing this?
Hi,
maybe your RTP stream is not getting through the asterisk box due to
canreinvite=yes setting in your SIP profile?
Nope :-(
less /etc/asterisk/sip.conf | grep canreinvite
canreinvite=no
canreinvite=no
canreinvite=no
canreinvite=no
canreinvite=no
canreinvite=no
What is result of the following test in your dialplan?
exten = 123,1,NoOp(***INCOMING CALL***)
exten = 123,n,Set(CHANNEL(language)=en)
exten = 123,n,Answer()
exten = 123,n,Read(CONFNO,conf-getconfno,4)
exten = 123,n,Playback(conf-enteringno)
exten = 123,n,SayDigits(${CONFNO})
exten = 123,n,Hangup
As per my current problem.
SIPGATE CUSTOMER - SIPGATE - ASTERISK {WORKS}
@inbound-sipgate-584e;2 Playing 'conf-enteringno.gsm' (language
'en')
/PRESS 1234 AND READ BACK DETECTED WITHOUT ERROR/
-- Executing [...@cc-test:6] SayDigits(@inbound-sipgate-584e;2,
1234) in new stack
BUT - PSTN - SIPGATE is nogo:
PSTN - SIPGATE - ASTERISK {BROKEN}
SIP/sipgate-dv-0008 Playing 'conf-getconfno.gsm' (language 'en')
/MASH KEYS AS MANY TIMES AS YOU LIKE - NOTHING DETECTED/
-- User disconnected
Cheers
Joern
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Re: [asterisk-users] DAHDI and Analogue lines (UK)
On Fri, 2010-01-15 at 22:26 +, Gordon Henderson wrote: On Sat, 16 Jan 2010, Tzafrir Cohen wrote: On Fri, Jan 15, 2010 at 04:06:54PM +, Gordon Henderson wrote: Have an intersting issue whem migrating a site from Zap on 1.3 to DAHDI on 1.4.. Nothing special about the hardware - older TDM400 card, 2 red modules fitted... Both channels work fine under 1.2/Zaptel. With 1.4/DAHDI both channels still work OK, but only for one line - the 2nd line causes it to refuse to dial-out no matter which port it's plugged into. The Lines are bog-standard BT analogue lines and we're about 2Km from the exchange. Both sound good to me and dial out OK with a test phone connected to them, but only one will dial-out via the PBX. This is what I see: [Jan 1 05:14:14] WARNING[1200]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) And yet the line isn't busy or congested - nothing's using it. The output of dsx*CLI dahdi show status Description Alarms IRQbpviol CRC4 Wildcard TDM400P REV E/F Board 5 OK 0 0 0 is fine, as is: dsx*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 1incoming default 2incoming default So I'm a bit stuck. Why doesn't DAHDI like that particular line? What does it do to it that Zap didn't? What version of Zaptel? Oldish - Zaptel Version: 1.2.23 What is the value of 'InAlarm' from 'dahdi show channel 2' ? InAlarm: 1 That's not good, is it... Doesn't explain why an analogue phone connected to the line works OK though - or can it indicate another sort of fault, or is it just too fussy? The line itself is their FAX line, although I'm not using it for FAXes - just as a second outgoing call line (I have it arranged to innore incoming calls - which are detected) There is also another phone on the line, so 3 devices including the asterisk box, however I got the same result with it plugged directly into the master socket with nothing else connected. Gordon Just in passing Gordon - call that line from an external phone and see if the alarm clears. I've had some DAHDI issues where the alarm is up until the line takes an incomming call, but it still works. -- _ -- 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] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, 2010-01-13 at 10:12 -0500, Kristian Kielhofner wrote: On Wed, Jan 13, 2010 at 2:43 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: Thanks for that. Looking at the RTP packets I can see two types as you point out. The first appears to be the audio: Real-Time Transport Protocol 10.. = Version: RFC 1889 Version (2) Payload type: ITU-T G.711 PCMU (0) And as you say, the DTMF events are clear to see: RFC 2833 RTP Event Event ID: DTMF One 1 (1) ..00 1010 = Volume: 10 So, as these can be seen in the stream, do I need to tell Asterisk to detect these? Does it not do that when I set: dtmfmode=rfc2833 ??? There are some pretty widely recognized RFC2833 compatibility issues in the SIP/RTP world. I had a nasty feeling something like that was coming :-( Which version of Asterisk are you using? Asterisk 1.6.1.11 Do you know what kind of equipment your carrier is using? If they are using Asterisk you can try to add rfc2833compensate=yes to their peer entry in sip.conf. I've not been able to get that out of them, but I don't *think* It's Asterisk based because they say: Unfortunately, our assistance with Asterisk is extremely limited. For configuration problems you will have to rely on other sources. [http://www.sipgate.co.uk/faq/index.php?do=displayArticlearticle=540qw=asterisk] The SIP debug, however, will tell you if the remote end is configured to use RFC2833 or not. That's why I was telling you to look for telephone-event in the INVITE from your provider. Keep in mind SIP (most likely) runs over UDP between you and your provider, not TCP. I see a 'telephone-event' : a=rtpmap:101 telephone-event/8000 That's all you need to know. They are configured for RFC2833 and they're sending RFC2833. I appreciate this is a 'how long is a piece of string question Kristian, but is there likely to be a way I can fix this? -- _ -- 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] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, 2010-01-13 at 11:13 -0500, Kristian Kielhofner wrote: On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: ..snip.. I've not been able to get that out of them, but I don't *think* It's Asterisk based because they say: Unfortunately, our assistance with Asterisk is extremely limited. For configuration problems you will have to rely on other sources. [http://www.sipgate.co.uk/faq/index.php?do=displayArticlearticle=540qw=asterisk] Just because they don't offer assistance with Asterisk doesn't mean they don't use it themselves. If you send me a packet capture in PCAP format with SIP+RTP between your system and your carrier I can debug this further. That may contain sensitive data, such as SIP account/password details - so I'll pass on that, but thanks for the offer. -- _ -- 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] Sipgate DTMF not detected
On Tue, 2010-01-12 at 10:37 -0500, Kristian Kielhofner wrote:
On Mon, Jan 11, 2010 at 12:19 PM, Steve Howes steve-li...@geekinter.net
wrote:
Codec? I've had 2833 do funny things with anything other than a/ulaw
(might just be me though..)
S
--
Codecs other than G711u/a don't support inband DTMF. Seeing as INFO
is rarely used that pretty much leaves RFC2833. Turn on SIP debugging
and look in the INVITE from the provider for telephone-event. If you
see it, they're configured to use RFC2833.
If they are, you need to do a packet capture or other RTP debug to see
the out of band RFC2833 events.
--
Kristian Kielhofner
Assuming that I enable debugging using:
asterisk -rvv
CLI sip set debug on
Then with this:
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
I see nothing nothing showing keypresses scroll past me. Even a SIP TCP
dump shows nothing. SIPGATE have said;
you should be able to set the dtmfmode to rfc2833 in your default
sip.conf.
Best regards,
Frederik
I've tried other combinations such as info, inband et al. I'm guessing
{that's all it is} that rfc2833 will signal the dtfm over sip as opposed
to in the audio stream?
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Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Tue, 2010-01-12 at 16:52 -0500, Kristian Kielhofner wrote:
On Tue, Jan 12, 2010 at 12:09 PM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
Assuming that I enable debugging using:
asterisk -rvv
CLI sip set debug on
Then with this:
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
I see nothing nothing showing keypresses scroll past me. Even a SIP TCP
dump shows nothing. SIPGATE have said;
you should be able to set the dtmfmode to rfc2833 in your default
sip.conf.
Best regards,
Frederik
I've tried other combinations such as info, inband et al. I'm guessing
{that's all it is} that rfc2833 will signal the dtfm over sip as opposed
to in the audio stream?
RFC2833 is carried in RTP like the audio stream. However, it uses a
different payload type from the RTP packets used to transport the
audio. If you did an RTP capture you would be able to see the RFC2833
events (which correspond to DTMF presses).
Thanks for that. Looking at the RTP packets I can see two types as you
point out. The first appears to be the audio:
Real-Time Transport Protocol
10.. = Version: RFC 1889 Version (2)
Payload type: ITU-T G.711 PCMU (0)
And as you say, the DTMF events are clear to see:
RFC 2833 RTP Event
Event ID: DTMF One 1 (1)
..00 1010 = Volume: 10
So, as these can be seen in the stream, do I need to tell Asterisk to
detect these? Does it not do that when I set: dtmfmode=rfc2833
???
The SIP debug, however, will tell you if the remote end is configured
to use RFC2833 or not. That's why I was telling you to look for
telephone-event in the INVITE from your provider. Keep in mind SIP
(most likely) runs over UDP between you and your provider, not TCP.
I see a 'telephone-event' :
a=rtpmap:101 telephone-event/8000
buried in the chunk below. but I have to be honest, SIP is new to me so
I'm not sure of myself with this:
v=0
o=root 27089 27089 IN IP4 217.10.69.13
s=session
c=IN IP4 217.10.69.13
t=0 0
m=audio 19990 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
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[asterisk-users] Sipgate DTMF not detected
I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to recognize digits pressed on a keypad coming in from a Sipgate trunk. There answer was to set this: dtmfmode=rfc2833 in the general section of sip.conf This has made no difference. I've tried a range of settings (auto, rfc2833,info) but no matter what, it plain refuses to pick up key presses. Locally, if I call from an extension on an ata or a softphone, it works flawlessly (I have no fxo, everything is SIP based). It's extremely frustrating and I would be grateful if anyone could offer some help troubleshooting and fixing this? -- _ -- 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] Sipgate DTMF not detected
On Mon, 2010-01-11 at 16:52 +, Steve Howes wrote: On 11 Jan 2010, at 16:26, listu...@spamomania.co.uk wrote: This has made no difference. I've tried a range of settings (auto, rfc2833,info) but no matter what, it plain refuses to pick up key presses. snip It's extremely frustrating and I would be grateful if anyone could offer some help troubleshooting and fixing this? Try asking Sipgate what settings you should use? If they are sending it as audio, make sure you are using suitable codecs etc. Try SIP traces to see what you can see. Steve Steve, you've snipped the bit that said: There answer was to set this: dtmfmode=rfc2833 in the general section of sip.conf But thanks, been there and done that. -- _ -- 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] No dial-tone with X101P FXO card
On Sun, 2010-01-10 at 00:25 -0800, Nitin Bahadur wrote: Hi Tzafrir, Some more background...I have a comcast phone line which I have connected to my FXO port. When I call my number, it goes directly to comcast voicemailin other words, there is no ringing tone and pickup by asterisk. That would suggest the card is looping the line (busying it out). -- _ -- 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] Per user voicemail greeting
I'm struggle to answer a simple question. One user at extension 4000 wants a custom .gsm file to play for their mailbox. I can't figure where to put it/what to set in voicemail.conf to achieve this: voicemail.conf 4000 = 4000,system,voicem...@net Relevant extensions.conf line: exten = 2,n,VoiceMail(4...@voicemail) It all works fine, playing the system VM greating, but I would like to use the custom .gsm for this user only. Can anyone help? ___ -- 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] Per user voicemail greeting
On Wed, 2009-12-30 at 10:00 -0500, Doug Lytle wrote:
listu...@spamomania.co.uk wrote:
I'm struggle to answer a simple question. One user at extension 4000
wants a custom .gsm file to play for their mailbox. I can't figure where
to put it/what to set in voicemail.conf to achieve this:
And this custom .gsm file is a greeting? Or, you're talking about
custom prompts for that user?
It's a greeting converted from a .wav and what I'm looking to do is have
just this one box play this specific greeting.
It's not possible for the user to log in and change this - it's a
'system' mailbox with no client attached to it. The logic of which is a
bit odd (basically it's a voicebank only)
I don't mind duplicating the greeting into different formats - that's
not an issue. I just need to know where I put them for 4000.
For example, if I look at the default box '1234' I don't really get any
clues as the default playing message is called: vm-intro.gsm
ls -alh /var/spool/asterisk/voicemail/default/1234/
drwxr-xr-x 2 root root 4.0K 2009-12-09 07:47 en
drwxr-xr-x 2 root root 4.0K 2009-12-09 07:47 INBOX {EMPTY}
..
ls -alh /var/spool/asterisk/voicemail/default/1234/en/
total 32K
-rw-r--r-- 1 root root 8.7K 2009-12-09 07:47 busy.gsm
-rw-r--r-- 1 root root 8.6K 2009-12-09 07:47 unavail.gsm
What I need to know are the formats I need and location I need to
satisfy it unless there is some hideously simple 'Use this switch to
specify the greeting file' I'm missing?
ls -alh /var/spool/asterisk/voicemail/voicemail/4000/
-rw-r--r-- 1 root root 31K 2009-12-30 14:45 CUSTOMvm-intro.gsm
drwxr-xr-x 2 root root 4.0K 2009-12-30 14:46 INBOX
drwxr-xr-x 2 root root 4.0K 2009-12-30 14:46 tmp
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Re: [asterisk-users] SOLVED IN PART Per user voicemail greeting
On Wed, 2009-12-30 at 15:18 +, Ishfaq Malik wrote: Get the customer to log into their voicemail mailbox and follow the instructions to record an unavailable message (Options 0 then 1 if there are no messages I think) Then in the conf you need exten = 2,n,VoiceMail(4...@voicemail,u) Ish Thanks Ish, that's pretty much it. I skipped the first step, copied and renamed the custom greeting to: /var/spool/asterisk/voicemail/voicemail/4000/unavail.gsm Changed the extensions.conf as you said from: exten = 2,n,VoiceMail(4...@voicemail) To: exten = 2,n,VoiceMail(4...@voicemail,u) And away it goes :-) The only downside is it still plays the system default (vm-intro.gsm) *after* the custom ends. SIP/1000-0001 Playing '/var/spool/asterisk/voicemail/voicemail/4000/unavail.gsm' (language 'en') SIP/1000-0001 Playing 'vm-intro.gsm' (language 'en') But it's good enough for what I need - many thanks to all that took the time and trouble to respond. ___ -- 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] SIP Issue
On Mon, 2009-12-28 at 12:11 -0600, James A. Shigley wrote:
Alright I have a SIP phone located off premises with a very annoying
issue.
Well I say a sip phone it is actually two phones hooked to a Cisco Spa
2102
Link: http://www.cisco.com/en/US/products/ps10026/index.html
Looks pretty much like the PAP2 which I have running flawlessly with 1.6
in and outbound - so don't despair, you can solve this.
Each phone being a different line/extension.
Alright either line can ALWAYS make outbound calls no issue. The
problem is on the Inbound side. I’m completely stumped as to why. I
could make 10 back to back out bound calls and then call inbound and
watch the call come in to * and try to be passed to the sip phone only
to get “Error Message 14: Not a Working Number.” So it doesn’t seem to
be a matter of the phones Sip Login “Timing out”
And when I check sip peers it shows the correct IP address of the box
so it doesn’t appear to be that it can’t find the Cisco box.
Here is what I use for the inbound context, replacing the _X_ with the
actual extension of course.
[to_ddwhome]
exten= _X_,1,wait(1)
exten= _X_,n,Dial(${ddwhome},21)
exten= _X_,n,Goto(dial_inf,${EXTEN},1)
${ddwhome}=SIP/ddwhome
Now the odd thing is when it gets the Error 14 message then the third
step to dial_inf does not execute. Though when it rarely does connect
with the sip phone if no one answers in 21 seconds than it will roll
over to that step.
Any ideas?
James Shigley
Probably be useful to see sip.conf as well and know the version of
Asterisk you are running but in passing, you don't have any firewall
rules that could stop your asterisk talking TO the Cisco when something
comes in?
The one minor issue I had with mine is my router has some NAT issues
with signalling (It's a Draytek - they are known for it). In the end I
shifted the PAP2 up to 5061/5062 and the problem was gone. None of this
may be useful to you but I'll tell you this much. In my few weeks with
Asterisk I've had times where I've asked myself why certain things would
plain refuse to work and on every occasion it was *not* the fault of
Asterisk. 50% my config, 40% my network, 10% different docs for
different versions and missing info ;-)
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[asterisk-users] Q; Recording when a bypass phone is used
The answer is probably no, but I have a bypass PSTN phone ahead of my Asterisk 1.6 box. I noticed when a call is answered on this bypass phone, the 'record' option still partially operates on Asterisk, but stops after the ring detection goes low. Is it possible to have Asterisk record when a DAHDI channel is detected in the off hook state, even if this is outside of Asterisk? ___ -- 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] Recording the Calls to a USB Drive
On Thu, 2009-12-24 at 18:53 +0100, Gergo Csibra wrote:
Thursday, December 24, 2009, 5:41:46 PM, Danny wrote:
Just my opinion; unless you are recording long or many long calls, you
should record to your local drive, then copy the files to the USB drive.
Asterisk is a very good tool - you don't need to mess it up by introducing
an easy point of failure.
Yes. I do this since 3 years and work very well.
What would be the problem with mounting the usb disc somewhere like:
/mnt/usbdisk and using something like:
exten = s,2,MixMonitor(/mnt/usbdisc/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%
S)}-${UNIQUEID}.wav,v(0))
???
This should be good for anything capable of being mounted (smb, nfs et
al). That's one of the beautiful things about Linux. It does not care
what the device is - just that it can find it.
Of course, the caveat - if it's not mounted, it can't write - but I'm
sure the excellent developers of Asterisk have coded to catch basic
exceptions like 'file not found'.
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Re: [asterisk-users] 1.6 Troubleshooting help
Dave Wrote: It looks like whatever is being transmitted, or the response, isn't getting through. Possibly due to NAT or a firewall? It would help if you described the scenario where this is occurring. Indeed, my post was gibberish :-O This was a 'nat' issue, but not in the traditional sense. Draytek router getting it's knickers in a twist and not wanting to play happy sockets. ___ -- 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] 1.6 Troubleshooting help
Hi, How would I go about troubleshooting this: [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:13] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:14] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:15] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:16] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:19] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:20] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:22] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:24] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:26] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:28] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:30] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:32] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:35] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. ___ -- 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] SOLVED PAP2 Dialing Delay
On Sun, 2009-12-20 at 16:16 -0400, Tim Johnson wrote:
Possibly OT?
I've hooked up a Linksys PAP2 to my Asterisk 1.6 fairly painlessly. The
only issue I can't beat with it is the dial delay when calling internal
or external numbers.
No matter what it seems to take 10 -15 seconds to actually dial. I've
altered the device removing all *xx combos and unnecessary waffle and
cut the dialplan string to (x.S0) but the problem persists.
Anyone else seen this issue?
Have you adjusted the Interdigit Long Timer and Interdigit Short Timer in
the Regional menu?
Tim
It's related to that, but rather than break stuff changing it I found
the fix for this was the dialplan string. It seems to have a 'feature'
with the wildcard '.'. That is if you have x.S0 it will wait beyond the
end of Interdigit Long Timer (as it does not know how many digits you
may dial). Whereas if you specify (for the UK national numbers)
9xxxS0 it's fires straight out as expected.
Personally I would have expected it to have waited for the Interdigit
Short Timer before taking the S0 instruction - but that's probably down
to my understanding {or lack thereof ;-)} of what is going on.
I meant to follow this up for the archives - so here goes;
Linksys PAP2 users suffering from a 10-15 second delay on dialling out
with Asterisk can cure this problem by checking the Linksys Dialplan is
not making excessive use of the period '.' wildcard. Specify the correct
number of digits with 'x' followed by STRAIGHT OUT (S0) will cure this
behaviour. The dialplan string can be found by logging into the Linksys
as the admin (not user), switch to advanced mode, and select line 1/2 to
suit. It's towards the bottom of the page.
Lowering the Interdigit Long Timer and Interdigit Short Timer setting
can also have an impact on this, but may cause other issues with slow
dialling users.
This may (unchecked) also apply to Linksys IP phones that use the same
firmware/guts ???
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[asterisk-users] PAP2 Dialing Delay
Possibly OT? I've hooked up a Linksys PAP2 to my Asterisk 1.6 fairly painlessly. The only issue I can't beat with it is the dial delay when calling internal or external numbers. No matter what it seems to take 10 -15 seconds to actually dial. I've altered the device removing all *xx combos and unnecessary waffle and cut the dialplan string to (x.S0) but the problem persists. Anyone else seen this issue? ___ -- 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] dahdi-channels.conf -v- chan_dahdi.conf
Some recent issues I had with hardware seem to come back to not understanding two very similarly named files: /etc/asterisk/dahdi-channels.conf /etc/asterisk/chan_dahdi.conf I've modified the chan_dahdi.conf to work now, but it would appear all I needed to do was include dahdi-channels.conf in chan_dahdi.conf and the problem would not have persisted? Is it me or is that a bit Monty Python? /etc/asterisk/dahdi-channels.conf ; Autogenerated by /usr/sbin/dahdi_genconf on Sun Dec 13 18:13:02 2009 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) ;;; line=1 WCFXO/0/0 FXSKS (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 1 /etc/asterisk/chan_dahdi.conf [channels] usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 signalling=fxs_ks channel = 1 ___ -- 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] Is this bad hardware? Dahdi-v-X100 clone
I've spent a week playing with Asterisk 1.6 and I love it. What a brilliant piece of software! Progress and learning have been reasonably good. I have external SIP provider calls coming in and have put together a little call platform and I'm stunned at the flexibility. There is one issue for me. I took me a while to click that ZAPTEL now equals Dahdi, but now I'm there I have an issue with the a X100 clone card that I have been told *not* to mention as I'm guaranteed a hostile response :- So, I've put on my flameproof pants to ask a simple question: dahdi show status gives a red alarm. I'm guessing this means the card is unable to detect the battery. I've plugged a test but into the loop through on the card, dialtone is there. I've tried reversing the polarity, two way/three way jack leads (I'm in the UK) but none the less I get: Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 RED 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) Is this likely to be bad hardware (hostility towards this cheap card noted) or software/driver? It would be just ace to crack this problem and learn some more. Thanks Bob. ___ -- 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] Is this bad hardware? Dahdi-v-X100 clone
On Mon, 2009-12-14 at 14:17 -0300, Vinícius Fontes wrote: I have never used that card myself, but I have never seen an analog board reporting a RED alarm. Probably there is something incorrect in your configuration. Please post your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf. Vinícius Fontes www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP Here it is Vinicius, the only thing standing out to me is the card is UK but showing as US. The red alarm may be totally irrelevant - but I don't seem to be able to get it to work :-( less /etc/asterisk/chan_dahdi.conf | sed '/^\;/d' [trunkgroups] [channels] usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 less /etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Sun Dec 13 18:13:02 2009 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) fxsks=1 echocanceller=mg2,1 # Global data loadzone= us defaultzone = us ___ -- 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] Is this bad hardware? Dahdi-v-X100 clone
On Mon, 2009-12-14 at 11:11 -0600, Tilghman Lesher wrote: On Monday 14 December 2009 10:28:08 am listu...@spamomania.co.uk wrote: I've spent a week playing with Asterisk 1.6 and I love it. What a brilliant piece of software! Progress and learning have been reasonably good. I have external SIP provider calls coming in and have put together a little call platform and I'm stunned at the flexibility. There is one issue for me. I took me a while to click that ZAPTEL now equals Dahdi, but now I'm there I have an issue with the a X100 clone card that I have been told *not* to mention as I'm guaranteed a hostile response :- So, I've put on my flameproof pants to ask a simple question: dahdi show status gives a red alarm. I'm guessing this means the card is unable to detect the battery. I've plugged a test but into the loop through on the card, dialtone is there. I've tried reversing the polarity, two way/three way jack leads (I'm in the UK) but none the less I get: Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 RED 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) Is this likely to be bad hardware (hostility towards this cheap card noted) or software/driver? Getting dialtone on the passthrough port (it's passthrough, not loopback) doesn't tell you much, as the pins are usually hardwired between the two jacks. Sure. I agree. It does tell you the leads to that point are OK, but nothing more :-) Generally, what we tell people here who are having hardware problems are to contact their reseller for support. That's true, whether the cards are Digium, Sangoma, or a cheap clone. However, given that cheap clones generally have no support system, it's interpreted as hostility when we cannot offer any particular help. The 'hostility' line is in the Asterisk book. There is a warning on 'clone' cards that speaks of 'hostility'. Apologies for any offence. I don't want to start a war, but there is a square to that. I'm new to Asterisk having spent years in analogue telephony. If I can get a test Asterisk working on a cheap clone card without a hitch, I'm most likely to expand this and buy TDM400's and above and swear its virtues to all. However, as they cost of some of the DIGIUM cards is about the same (if not more) than many SIP gateways suitable for SOHO's and SME, I'm unlikely to buy an expensive hardware card just to 'prove' it works OK on a whim. I fully understand the point you make and the thinking behind it. I can understand the commercial and revenue protection angle. It's no big deal, I have no need to make it work and there are other projects I can be studying. Thanks anyway. ___ -- 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] Is this bad hardware? Dahdi-v-X100 clone
On Mon, 2009-12-14 at 14:10 -0600, Tilghman Lesher wrote: I don't want to start a war, but there is a square to that. I'm new to Asterisk having spent years in analogue telephony. If I can get a test Asterisk working on a cheap clone card without a hitch, I'm most likely to expand this and buy TDM400's and above and swear its virtues to all. On the contrary, you're likely to continue to buy clone cards. If there's no advantage to buying premium hardware, why would anybody spend the excess cash? The quality argument springs to mind - but it depends on the final application. Perhaps the DIGIUM cards are priced unrealistically for what they are? I can't really comment on the commercial aspects as it's not an area where I have sufficient qualification. However, as they cost of some of the DIGIUM cards is about the same (if not more) than many SIP gateways suitable for SOHO's and SME, I'm unlikely to buy an expensive hardware card just to 'prove' it works OK on a whim. Actually, many people have taken it a step further than that. If you can get a SIP trunk provider on a broadband connection who will provision a telephone number to you, why are you even bothering with analog telephony at all? It's a valid point. Sometimes it's about bringing together what you already have on site, without putting it out to a farm. There is also the resilience and multiple points of failure angle. That said, I've seen a business crippled when a Shortel system failed. The ISDN 30 on fibre was useless without the hardware. In this case the saving grace was a PSTN faxline with DSL. This was able to pipe calls diverted to an ITSP (Sipgate) for collection. The square would be if you have an ISTP remotely handling everything, their failure can cripple you, as can a (remarkably common) DSL failure. But yes, it's a serious option suitable to some, in the same way BT Featureline Compacts suited some :-) ___ -- 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] Is this bad hardware? Dahdi-v-X100 clone
On Mon, 2009-12-14 at 22:27 +0200, Tzafrir Cohen wrote: On Mon, Dec 14, 2009 at 02:17:39PM -0300, Vinícius Fontes wrote: I have never used that card myself, but I have never seen an analog board reporting a RED alarm. Ahem. Wcfxo always has (AFAIR). Red alarm means that no line is connected (it gets no current from the remote FXS in the central office). Later on most DAHDI drivers of FXO ports started to use this signalling (though for the specific port rather than for the whole span). If you connect a standard analog phone instead of that card and get a dial tone, something is fishy with the card (or the driver) as it is mis-reporting. It turned out to be an issue with DAHDI. Once the signalling was added to the conf and an incoming call was made, the status shifted to OK. This may suggest the driver was failing to detect the line status from the card - but as it is a 'clone/unsupported' there is little point being concerned about it :-) ___ -- 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
