Re: [asterisk-users] Caller ID in Asterisk
Hi All,
Finally I am able to get the number displayed at the SIP side using
exten => _988.,1,Set(CALLERID(num)=8001234000)
exten => _988.,n,Dial(DAHDI/g1/${EXTEN},20)
However this number is fixed and I want to display the number of the
individual lines whoever is calling. I tried with
exten => _988.,1,Set(CALLERID(num)=${exten}) and exten =>
_988.,1,Set(CALLERID(num)=${EXTEN})
Both the above lines didn't help.
I have 8 lines configured as below and need the callerID of the
individual lines to be displayed at the SIP side
exten => 8001234001,n,Dial(DAHDI/32,,rt)
exten => 8001234002,n,Dial(DAHDI/33,,rt)
exten => 8001234003,n,Dial(DAHDI/34,,rt)
exten => 8001234004,n,Dial(DAHDI/35,,rt)
exten => 8001234005,n,Dial(DAHDI/36,,rt)
exten => 8001234006,n,Dial(DAHDI/37,,rt)
exten => 8001234007,n,Dial(DAHDI/38,,rt)
exten => 8001234008,n,Dial(DAHDI/39,,rt)
Warm Regards
Warm Regards
Venugopal G
HNM-SO WiMAX CPE VoIP IOT Team
Cell : +91-99723-99437
*
From: Gopalakrishnaiyer Venugopal-Q16770
Sent: Thursday, March 04, 2010 6:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID in Asterisk
Hi Jimmy,
Appreciate your help.
I tried the one below and cudnt get the caller ID.I am getting "Private
Call" and "Out of Area" in the sip phone display when i call from
asterisk.
My current extensions.conf looks like below
[general]
static=yes
writeprotect=no
autofallthrough=no
extenpatternmatchnew=no
clearglobalvars=no
priorityjumping=yes
userscontext=default
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel
username/password
;IAXINFO=myuser:mypass
TRUNK=DAHDI/G1
TRUNKMSD=1
[Internal]
include => Incoming
exten => 8001234001,1,Dial(DAHDI/32,,rt)
exten => 8001234002,1,Dial(DAHDI/33,,rt)
exten => 8001234003,1,Dial(DAHDI/34,,rt)
exten => 8001234004,1,Set(CALLERID(num)=8001234004)
exten => 8001234004,n,Set(CALLERID(name)="Line 4")
exten => 8001234004,3,Dial(DAHDI/35,,rt)
exten => 8001234005,1,Dial(DAHDI/36,,rt)
[Incoming]
exten => s,1,Answer
exten => s,2,Dial(DAHDI/g1,20,rt)
exten => _988.,1,Dial(DAHDI/g1/${EXTEN},20)
I also tried changing the dial plan to exten =>
_988.,3,Dial(DAHDI/g1/${EXTEN},20) and in that case the call itself was
not going through
Venugopal
From: asterisk-users-boun...@lists.digium.com on behalf of Jimmy Godbout
Sent: Thu 3/4/2010 5:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID in Asterisk
Hi,
You need to set the callerid before making the call, not after. Also, I
guess it's a typo that the priority in this dialplan is all 1; it should
be
exten => 8001234003,1,Set(CALLERID(num)=8001234003)
exten => 8001234003,n,Set(CALLERID(name)="Line 5")
exten => 8001234003,n,Dial(DAHDI/34,,rt)
Unless your using variable for the name and the number, you should not
put them in ${}.
Jimmy
-Original Message-
From: venui...@motorola.com
Sent: Thu, 4 Mar 2010 19:50:03 +0800
To: asterisk-users@lists.digium.com,
asterisk-users@lists.digium.com, asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten => 8001234003,1,Dial(DAHDI/34,,rt)
exten => 8001234003,1,Set(CALLERID(num)=${8001234003})
exten => 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not
registered.
Kindly help me...
From: asterisk-users-boun...@lists.digium.com on behalf of
Gopalakrishnaiyer Venugopal-Q16770
Sent: Thu 3/4/2010 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
Subject: [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the
SIP server.When i call from the Asterisk machine to the SIP server i am
not getting the caller id of the lines at the sip side.
Please help me to identify how this can be set.The
extensions.conf file is attached.
Cheers
venu
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Re: [asterisk-users] SIP / Echo Cancellation
Very informative post Vinícius ! 2010/3/5 Vinícius Fontes > - "Chandrakant Solanki" escreveu: > > > Hello > > > > I have successfully compiled OSLEC for echo cancellation for DAHDI > > channel. > > > > Is there any way to do echo cancellation for SIP Channel. > > > > Is any, please suggest me.?? > > > > Thanks in advance.. > > > > -- > > Regards, > > > > Chandrakant Solanki > > Short answer: Maybe. Depends on the SIP device you're using. > > Long answer: > *takes a deep breath* > > First you gotta understand why echo occurs. Every single call you've ever > made on your life has echo. You can hear yourself when you're speaking. If > that was not the case, it would feel like talking on a push-to-talk system. > So echo is a natural and even desirable phenomenom. What makes echo > unconfortable is when the echo is *delayed* too much. > > There's a number of causes for this to happen. First and foremost, > sometimes a part of the signal you're transmitting is reflected back to you. > That usually happens on the analog part of the system (analog phones as a > whole, the handset of an IP phone, the headset connected to your computer's > sound card, etc). When we're talking about VoIP, the latencies involved are > much higher than a completely TDM system. There's the encoding latency, > easily understood as the time the device takes to convert the analog signal > (your voice) in RTP packets, then there's the transmission latency, inherent > to any network, and so on. All those latencies add up to each other, making > the total latency go skyhigh and making you hear your own voice delayed by > some milisseconds - the infamous echo. > > Asterisk cannot cancel echo when the call is entirely IP, from an IP phone > to another, for example. There's simply no need for that. That's because > it's the device's job to cancel the echo caused by its own TX reflections or > analog/digital conversions. On the other hand, Asterisk can and will cancel > echo if you have a hardware echo canceller or a software based one, like > OSLEC -- which is by far the best software echo canceller I've ever seen. > > Finally, in order to solve your problem, you'll need to check a few things. > If the call is entirely VoIP, from one end to other, then the IP phones, > ATAs, gateways, softphones, whatever, are the sole responsibles on > cancelling the echo. You'll need to turn on echo cancelling on this devices > or tweak its parameters. Also, don't forget that latency makes echo much > worse. If you control the entire network between the two phones, you MUST > set up a QoS policy in order to minimize the latency as much as possible. > I've solved many echo problems by just implementing end-to-end QoS on the > network. > > Lastly (I swear I'm finishing this essay right here :), if that's not your > case and you're having echo issues calling from a SIP phone to an external > number, double check if OSLEC is indeed set as the echo canceller on > /etc/dahdi/system.conf and enabled with echocancel=yes on your > chan_dahdi.conf. You can always check if the echo canceller is active on a > certain DAHDI channel by issuing the command "dahdi show channel XX" on > Asterisk CLI, where XX of course is the said DAHDI channel. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec translation in Asterisk
Nobody to take this one! Am I missing anything in knowing following issue? --Hi Group, --Can anybody explain me in detail how the codec translation happens on --asterisk side when 2 endpoints have different codecs? --Thanking you in advance. SM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 30 mins GSM file
On Thursday 04 March 2010 16:51:54 David @ULC wrote: > I need to create 30 mins of GSM file for Asterisk . > > Silent / Blank file. > > Whats the best way to create it ? One of the nicest things about gsm files is that having no file header, you can concatenate multiple files and get the same effect as having played the series of files. Within the standard set of files is silence/10, which is 10 seconds of silence. Concatenate 180 instances of that file, and the result will be 1800 seconds (30 minutes) of silence. for i in `seq 1 180`; do cat /var/lib/asterisk/sounds/en/silence/10.gsm >> /var/lib/asterisk/sounds/30-minutes-of-silence.gsm ; done -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Agents
I've got a ton of files in doc but not that file. On Thu, Mar 4, 2010 at 7:32 PM, Leif Madsen wrote: > Matt wrote: > > Already found it -- but I was under the impression this was deprecated > > and removed in 1.6? > > Try looking in the doc/ subdirectory of your Asterisk 1.6.2 source. You're > looking for the building_queues.txt file. > > Leif. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP / Echo Cancellation
- "Chandrakant Solanki" escreveu: > Hello > > I have successfully compiled OSLEC for echo cancellation for DAHDI > channel. > > Is there any way to do echo cancellation for SIP Channel. > > Is any, please suggest me.?? > > Thanks in advance.. > > -- > Regards, > > Chandrakant Solanki Short answer: Maybe. Depends on the SIP device you're using. Long answer: *takes a deep breath* First you gotta understand why echo occurs. Every single call you've ever made on your life has echo. You can hear yourself when you're speaking. If that was not the case, it would feel like talking on a push-to-talk system. So echo is a natural and even desirable phenomenom. What makes echo unconfortable is when the echo is *delayed* too much. There's a number of causes for this to happen. First and foremost, sometimes a part of the signal you're transmitting is reflected back to you. That usually happens on the analog part of the system (analog phones as a whole, the handset of an IP phone, the headset connected to your computer's sound card, etc). When we're talking about VoIP, the latencies involved are much higher than a completely TDM system. There's the encoding latency, easily understood as the time the device takes to convert the analog signal (your voice) in RTP packets, then there's the transmission latency, inherent to any network, and so on. All those latencies add up to each other, making the total latency go skyhigh and making you hear your own voice delayed by some milisseconds - the infamous echo. Asterisk cannot cancel echo when the call is entirely IP, from an IP phone to another, for example. There's simply no need for that. That's because it's the device's job to cancel the echo caused by its own TX reflections or analog/digital conversions. On the other hand, Asterisk can and will cancel echo if you have a hardware echo canceller or a software based one, like OSLEC -- which is by far the best software echo canceller I've ever seen. Finally, in order to solve your problem, you'll need to check a few things. If the call is entirely VoIP, from one end to other, then the IP phones, ATAs, gateways, softphones, whatever, are the sole responsibles on cancelling the echo. You'll need to turn on echo cancelling on this devices or tweak its parameters. Also, don't forget that latency makes echo much worse. If you control the entire network between the two phones, you MUST set up a QoS policy in order to minimize the latency as much as possible. I've solved many echo problems by just implementing end-to-end QoS on the network. Lastly (I swear I'm finishing this essay right here :), if that's not your case and you're having echo issues calling from a SIP phone to an external number, double check if OSLEC is indeed set as the echo canceller on /etc/dahdi/system.conf and enabled with echocancel=yes on your chan_dahdi.conf. You can always check if the echo canceller is active on a certain DAHDI channel by issuing the command "dahdi show channel XX" on Asterisk CLI, where XX of course is the said DAHDI channel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI and 1.6.1
I'm having this EXACT same problem, I haven;t been able to narrow down the cause of it yet, but it seems to me that users are receiving notifications for voicemails in mailboxes that belong to other people, as sometimes their mail count magically disappears, which I have been suspecting is when somebody else checks their VM. I found the problem also exists in 1.6.2 which is where I first noticed it (upgraded from 1.4.x to 1.6.2.x). I tried downgrading to 1.6.1 and the problem seemed not quite as bad, but I know its still present. I was actually quite surprised to find that nobody had previously mentioned the problem on this list when I came across it so I thought it might of been something specific to my situation. Even if you turn the polling options back on in the voicemail conf file the problem still persists. We are using all Aastra phones - a mix of 9133i, 9112i, 480, 35i, 57i phones - but the problem seem unrelated to the make/model of the phone based on seeing you having the same problem with Polycom's. Not sure that it should matter, but we are using FreePBX 2.6 ontop of asterisk and running it in "users and devices" mode (as apposed to the default "extensions" mode). If you do a voicemail show users from the Asterisk console it shows the correct VM counts for the mailboxes, so its not that Asterisk is counting them incorrectly, it just seems to be sending the notifications of VMs to the wrong places. I'm suddenly very glad I;m not alone on this one! I;m more than happy to do any testing of patches if anybody has any suggestions. -- Matt On Tue, Mar 2, 2010 at 1:36 PM, Dave Poirier wrote: > We are having an issue with Asterisk 1.6.1 and the MWI turning on when a > user doesn't have voicemail. We see random MWI lights come on and the phone > indicates a random number of messages (its been anywhere from 1-14) when a > server reload is done. > > I just checked one user, they have no messages old or new and the phone > (Polycom IP330) indicates that they have 2 messages. The user will check for > messages, the system will tell them that they have none and the light goes > out. > > I know that starting in 1.6 Asterisk moved from a polling system to an > event based system but it's unclear to me what is causing these events to be > generated. Anyone else experience this? Any tips, suggestions? > > Thanks, > Dave > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cli_originate malfunction after upgrade from 1.6.2.0 to 1.6.2.1-5
Tzafrir Cohen wrote: > IIRC this issue is fixed in latest SVN, and also in 1.2.6.3-rc2 (1.2.6.5 > is based on 1.2.6.2). Also I just finished releasing several new release candidates which should have the fix as well if it is indeed resolved. See the release announcement for the next set of release candidates at http://www.asterisk.org/node/49915 Thanks! Leif Madsen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Agents
Matt wrote: > Already found it -- but I was under the impression this was deprecated > and removed in 1.6? Try looking in the doc/ subdirectory of your Asterisk 1.6.2 source. You're looking for the building_queues.txt file. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan reload: not working with large dialplans
Andreas Brodmann wrote: > the dialplan currently holds 1792 lines. It's a plain old .conf file. That's interesting, because I have a dialplan over 2400 lines and it seems to load fine... However, I'm using this on an ABE machine which is based on 1.4. Perhaps I'll try loading this on a 1.6.2 box tomorrow. Leif! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PHPAGI and Asterisk 1.6
I am developing a few AGI programs using PHPAGI. This is the first time developing for Asterisk 1.6 and I am having a lot of problems reading variables with the $agi->get_variable construct. While an AGI debug shows me that I am asking for the variable and Asterisk is giving me the correct value, most of the time the PHP variable is empty or contains the wrong information. I never saw this problem with prior versions of Asterisk so I am wondering what the problem with the phpagi class could be? These scripts are short and for low usage systems so I do not bother writing them in C or other compiled language. Is anyone having the same problem? So far my workaround is to do a loop of $agi->get_variable until my php variable is not empty but that is ugly and wastes time. Since PHPAGI has not been updated in several years is there another PHP library for AGI development or is it better to try and build my own? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 30 mins GSM file
On Thu, 2010-03-04 at 23:27 +, Steve Howes wrote: > On 4 Mar 2010, at 23:11, Steve Edwards wrote: > > On Thu, 4 Mar 2010, Steve Edwards wrote: > >> On Fri, 5 Mar 2010, David @ULC wrote: > >> > >>> I need to create 30 mins of GSM file for Asterisk . > >>> > >>> Silent / Blank file. > >>> > >>> Whats the best way to create it ? > >> > >> Record yourself thinking of the solution for 1/2 of an hour. > > > > Use sox to concatenate 6.9 copies of John Cage's 4'33" > Was thinking about recomending recording the speeches of one of our managers. But then again, i realized that is just white noise ;-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Agents
Steve, Already found it -- but I was under the impression this was deprecated and removed in 1.6? On Thu, Mar 4, 2010 at 6:44 PM, Steve Edwards wrote: > On Thu, 4 Mar 2010, Matt wrote: > > > I'm trying to setup a situation where I have agents on POTS lines at > remote > > locations. I want to allow them to call a DID, log into the Asterisk > > system, and be an agent. Ultimately I'd like Asterisk to call them at > the > > number they were at when they logged in. > > > > Does this functionality exist in Asterisk? > > Google for AgentCallbackLogin. > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Agents
On Thu, 4 Mar 2010, Matt wrote: > I'm trying to setup a situation where I have agents on POTS lines at remote > locations. I want to allow them to call a DID, log into the Asterisk > system, and be an agent. Ultimately I'd like Asterisk to call them at the > number they were at when they logged in. > > Does this functionality exist in Asterisk? Google for AgentCallbackLogin. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 30 mins GSM file
On Thu, 2010-03-04 at 23:27 +, Steve Howes wrote: > On 4 Mar 2010, at 23:11, Steve Edwards wrote: > > On Thu, 4 Mar 2010, Steve Edwards wrote: > >> On Fri, 5 Mar 2010, David @ULC wrote: > >> > >>> I need to create 30 mins of GSM file for Asterisk . > >>> > >>> Silent / Blank file. > >>> > >>> Whats the best way to create it ? > >> > >> Record yourself thinking of the solution for 1/2 of an hour. > > > > Use sox to concatenate 6.9 copies of John Cage's 4'33" > Download Audacity and create a new track, then insert 30 minutes of silence. Save your wav and then convert to gsm with sox. sox foo.wav foo.gsm Done in seconds. Just make sure you save the wav file in mono and 8khz or you will have to resample the gsm file. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote Agents
I'm trying to setup a situation where I have agents on POTS lines at remote locations. I want to allow them to call a DID, log into the Asterisk system, and be an agent. Ultimately I'd like Asterisk to call them at the number they were at when they logged in. Does this functionality exist in Asterisk? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 30 mins GSM file
On 4 Mar 2010, at 23:11, Steve Edwards wrote: > On Thu, 4 Mar 2010, Steve Edwards wrote: >> On Fri, 5 Mar 2010, David @ULC wrote: >> >>> I need to create 30 mins of GSM file for Asterisk . >>> >>> Silent / Blank file. >>> >>> Whats the best way to create it ? >> >> Record yourself thinking of the solution for 1/2 of an hour. > > Use sox to concatenate 6.9 copies of John Cage's 4'33" Get permission first.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 30 mins GSM file
I believe we GSM of 8 bit for Asterisk ? On Fri, Mar 5, 2010 at 4:35 AM, David @ULC wrote: > Record a muted channel for 30 minutes like this: > > exten => s,1,Answer(1) > > exten => s,n,Progress() > > exten => s,n,record(silence_long.gsm|1800|s) > > exten => s,n,hangup > > > > > Above option looks easy. > > What I have to dial from soft phone to get this ? > > > > On Fri, Mar 5, 2010 at 4:21 AM, David @ULC wrote: > >> >> I need to create 30 mins of GSM file for Asterisk . >> >> Silent / Blank file. >> >> Whats the best way to create it ? >> >> >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 30 mins GSM file
On Thu, 4 Mar 2010, Steve Edwards wrote: > On Fri, 5 Mar 2010, David @ULC wrote: > >> I need to create 30 mins of GSM file for Asterisk . >> >> Silent / Blank file. >> >> Whats the best way to create it ? > > Record yourself thinking of the solution for 1/2 of an hour. Use sox to concatenate 6.9 copies of John Cage's 4'33" -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time/date over POTS?
On Thu, Mar 4, 2010 at 1:00 PM, Dave Fullerton < dfullertaster...@shorelinecontainer.com> wrote: My Linksys PAP2T-NA at home has it's own clock / NTP settings that sends the timestamp out to my analog phones. Check through the settings tab on your Linksys for a time setting. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 30 mins GSM file
Record a muted channel for 30 minutes like this: exten => s,1,Answer(1) exten => s,n,Progress() exten => s,n,record(silence_long.gsm|1800|s) exten => s,n,hangup Above option looks easy. What I have to dial from soft phone to get this ? On Fri, Mar 5, 2010 at 4:21 AM, David @ULC wrote: > > I need to create 30 mins of GSM file for Asterisk . > > Silent / Blank file. > > Whats the best way to create it ? > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 30 mins GSM file
On Fri, 5 Mar 2010, David @ULC wrote: > I need to create 30 mins of GSM file for Asterisk . > > Silent / Blank file. > > Whats the best way to create it ? Record yourself thinking of the solution for 1/2 of an hour. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 30 mins GSM file
- "David @ULC" wrote: > >I need to create 30 mins of GSM file for Asterisk . >Silent / Blank file. >Whats the best way to create it ? > Enable recording using monitor() or mixmonitor() in GSM format, call, then put your handset on mute for 30 minutes. :-) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 30 mins GSM file
Record a muted channel for 30 minutes like this: exten => s,1,Answer(1) exten => s,n,Progress() exten => s,n,record(silence_long.gsm|1800|s) exten => s,n,hangup _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC Sent: Thursday, March 04, 2010 4:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 30 mins GSM file I need to create 30 mins of GSM file for Asterisk . Silent / Blank file. Whats the best way to create it ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 30 mins GSM file
I need to create 30 mins of GSM file for Asterisk . Silent / Blank file. Whats the best way to create it ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] InterPBX communication using SIP
Hi Guys,
i am using the following config in pbx1:
register => pbx1:endop...@172.16.200.175
[pbx2]
type=friend
host=dynamic
trunk=yes
sercret=password
context=[default]
deny=0.0.0.0/0.0.0.0
permit=172.16.200.175/255.255.255.128
in pbx2:
register => pbx2:endop...@172.16.200.176
[pbx1]
type=friend
host=dynamic
trunk=yes
sercret=password
context=[default]
deny=0.0.0.0/0.0.0.0
permit=172.16.200.176/255.255.255.128
and i get the following in pbx1:
-- Executing [18...@default:1] Dial("SIP/8029-b7413678",
"SIP/pbx2/8021||TWw") in new stack
-- Called pbx2/8021
[Mar 4 16:49:13] WARNING[3392]: chan_sip.c:12679 handle_response_invite:
Received response: "Forbidden" from '"Khalid Touati" <
sip:8...@172.16.200.176 >;tag=as1dcf5ff2'
-- SIP/pbx2-09cf4468 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/8029-b7413678' status is 'CONGESTION'
though i am using the same config in IAX and it's working fine, also it's in
the same context (so i believe it's a context issue).
--
Abdullah
--
_
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Re: [asterisk-users] time/date over POTS?
Jeff LaCoursiere wrote: > On Thu, 4 Mar 2010, Dave Fullerton wrote: > >> Jeff LaCoursiere wrote: >>> I had a customer ask me about time/date information being sent to his >>> analog (attached to a Linksys SPA2102) answering machine. I didn't know >>> that POTS could carry this information. Is this something Asterisk could >>> send over SIP? >>> >>> Cheers, >>> >>> j >>> >> Time and date info on a POTS line is part of the caller ID stream. It is >> up to the analog endpoint sending the caller ID stream to know the >> current time to send. Anything that works with SIP should also have NTP >> capabilities and should be getting its time using that. >> >> -Dave >> > > Aha. Sadly I know that the incoming calls from our PSTN provider (over > RBS T1) do NOT carry caller ID, so what we are passing on via SIP to the > Linksys box must also be missing the time info. > > Is there any way to add that to the outgoing call to the Linksys box? > > Cheers, > > j The time and date in your case is being generated (or should be) by the sipura, not whatever is sending the call to the sipura. Time and date information is not included in SIP caller ID (to my knowledge). It's up to the SIP endpoint to know what time it is. Check the NTP settings on the sipura to make sure it is syncing its time with an internet time server. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time/date over POTS?
On Thu, 4 Mar 2010, Dave Fullerton wrote: > Jeff LaCoursiere wrote: >> I had a customer ask me about time/date information being sent to his >> analog (attached to a Linksys SPA2102) answering machine. I didn't know >> that POTS could carry this information. Is this something Asterisk could >> send over SIP? >> >> Cheers, >> >> j >> > Time and date info on a POTS line is part of the caller ID stream. It is > up to the analog endpoint sending the caller ID stream to know the > current time to send. Anything that works with SIP should also have NTP > capabilities and should be getting its time using that. > > -Dave > Aha. Sadly I know that the incoming calls from our PSTN provider (over RBS T1) do NOT carry caller ID, so what we are passing on via SIP to the Linksys box must also be missing the time info. Is there any way to add that to the outgoing call to the Linksys box? Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware
Hi, Im one of the user for this card. It works like charm. in my country i have to set the signalling to fxs_ls and it works. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Audio on pstn call
Hi IRFAN, Thanks for that, actually, I think my FXO card already struck by lightning. I;ve changed to another card, and now work like charm. - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware
On 4 Mar 2010, at 17:22, Aditya Kumar wrote: > I saw this in ebay. > only 1 FXO. > > Asterisk X100P(B2) FXO PCI For IP-PBX From U.S > link is : > http://cgi.ebay.com/Asterisk-X100P-B2-FXO-PCI-For-IP-PBX-From-U-S_W0QQitemZ160331750263QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item2554845b77#ht_3934wt_1165 > > This is just 14$.. > looks like the distributer sold >600 pieces. > http://www.voip-info.org/wiki/view/X100P+clone Might not be worth the $14.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware
Thanks Again steve . Actually I feel that is expensive for my initial requirement of maling Asterisk and Zaptel work on my Linux box. I saw this in ebay. only 1 FXO. Asterisk X100P(B2) FXO PCI For IP-PBX From U.S http://cgi.ebay.com/Asterisk-X100P-B2-FXO-PCI-For-IP-PBX-From-U-S_W0QQitemZ160331750263QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item2554845b77#ht_3934wt_1165 This is just 14$.. looks like the distributer sold >600 pieces. Did any one try this with Astersik? how did it work with the ZAPTEL(DHADI)?? were there any issues pl let me know... if it works than I want t order that :-) I am in USA west coast. if u know any one else who can give working pieces f a better deal please let me know. From: Steve Edwards To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sun, February 28, 2010 10:13:13 AM Subject: Re: [asterisk-users] Hardware On Sun, 28 Feb 2010, Aditya Kumar wrote: > Can any one please suggest me a Card which is economical.. > My requirement is one FXO and one FSO. (FXS) > Also, as steve suggested I cannot use ATA because > the out put to ATA-SPA is SIP. > > I want to make use of DHAHII(interface) so looking f card I've never used this vendor and I don't know which corner of the world you're in, but this seems like a pretty good deal: http://www.cetusvoip.com/product_info.php?cPath=1_18_19&products_id=2780 Digium TDM411B 1FXS / 1FXO Analog TDM PCI Card for US$220. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 link is : -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK CallerID -v- Wildcard W100P
> > Given the PSTN rings once a month if that - all the rest is SIP via > > ITSP, I can't justify the cost of 5 times a telephone for a card - but > > thanks for the heads up GK old chap. > > > It's not a big issue - but an irritation if anything. I suspect a £2.50 > > callerID unit, a serial IC and a soldering iron may end up being the > > solution :-) > > Good luck then - but do make sure the line is actually presenting caller > ID first! I had issues with some telewest lines until I ported them into > VoIP... Looks like it's possible to modify some of the old BT Caller Display units with nothing more than about a fivers worth of parts all in. I'll put it in the 'rainy day' project list. If PSTN was more of an issue I'd go with the card you suggest - it's just not a big enough deal to worry with it - but thanks. Really appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK CallerID -v- Wildcard W100P
Brian wrote: > At the risk of being flamed > > Has anyone had any success get the 'El cheapo' Wildcard W100P > clone's (£20 flavour) to work with UK Caller ID? > > I'm not sure what the status of Asterisk 1.6 is with respect > to UK caller ID, being that we have an odd method of sending > the FSK ahead of the ring, but I'm guessing I can't be the > first to ask this? Nope, I asked some years ago :) I could never get my Wildcard clone to work with UK CLID, no matter what patches I applied. I gave up in the end & implemented a very roundabout solution using a Pace modem, second computer, and a database... It worked, albeit a little slowly. > > Keeping in mind that cost is the most important factor, my > searches I've found a couple of suggestions - the most > promising of which was reading the CID from a serial modem. > However, I've tried a couple - on of which is a BT Enabler > that no amount of AT commands can get to give up the CID > - and concluded that the chances of finding a compatible > modem are probably slimer than getting the clone to work. There are a very small number of modems which work with CLID. Pace being the only ones that I know of, which worked reliably... and Pace went bust years ago. You can pick up 2nd hand Pace modems off eBay, but by the time you've done that, you may as well have bought an A400P card... which will do UK CLID out of the box. If you want to take the Modem route, send me a mail off-list, as I have an implementation that may work for you. > > Has anyone been able to get the cheap clone cards to offer > CID in the UK? Only the A400P. But, TBH, at £55 with 1xFXO module, that's pretty cheap these days. I can heartily recommend them for being a) more reliable, and b) quicker at CLID detection than the modem option... HTH. Ade. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time/date over POTS?
Jeff LaCoursiere wrote: > I had a customer ask me about time/date information being sent to his > analog (attached to a Linksys SPA2102) answering machine. I didn't know > that POTS could carry this information. Is this something Asterisk could > send over SIP? > > Cheers, > > j > Time and date info on a POTS line is part of the caller ID stream. It is up to the analog endpoint sending the caller ID stream to know the current time to send. Anything that works with SIP should also have NTP capabilities and should be getting its time using that. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK CallerID -v- Wildcard W100P
On Thu, 4 Mar 2010, Brian wrote: (your email is a bit weird - my clients not quoting it right )-: On Thu, 2010-03-04 at 15:35 +, Gordon Henderson wrote: On Thu, 4 Mar 2010, Brian wrote: > At the risk of being flamed > Has anyone had any success get the 'El cheapo' Wildcard W100P clone's > (£20 flavour) to work with UK Caller ID? Looks like there is a patch to Zaptel to make it work: http://www.voip-info.org/wiki/view/Asterisk+and+UK+Caller+ID but who knows if it's in & working... I hear that it does with Zaptel - but Asterisk 1.6 - DAHDI ;-) So use 1.2 and Zaptel then... Works for me. I have no reason not desire to even think about 1.6 right now... There are 1000's other still sticking to 1.2... I have ventured into 1.4 territory though, but oddly enough, it doesn't yet give me as secure a feeling as 1.2 does... http://www.voipon.co.uk/openvox-a400p01-p-669.html Given the PSTN rings once a month if that - all the rest is SIP via ITSP, I can't justify the cost of 5 times a telephone for a card - but thanks for the heads up GK old chap. It's not a big issue - but an irritation if anything. I suspect a £2.50 callerID unit, a serial IC and a soldering iron may end up being the solution :-) Good luck then - but do make sure the line is actually presenting caller ID first! I had issues with some telewest lines until I ported them into VoIP... Gordon-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK CallerID -v- Wildcard W100P
On Thu, 2010-03-04 at 15:35 +, Gordon Henderson wrote: > On Thu, 4 Mar 2010, Brian wrote: > > > At the risk of being flamed > > > Has anyone had any success get the 'El cheapo' Wildcard W100P clone's > > (£20 flavour) to work with UK Caller ID? > > Looks like there is a patch to Zaptel to make it work: > >http://www.voip-info.org/wiki/view/Asterisk+and+UK+Caller+ID > > but who knows if it's in & working... I hear that it does with Zaptel - but Asterisk 1.6 - DAHDI ;-) > > > I'm not sure what the status of Asterisk 1.6 is with respect to UK > > caller ID, being that we have an odd method of sending the FSK ahead of > > the ring, but I'm guessing I can't be the first to ask this? > > We do a line polarity reversal, then FSK the caller ID, then ring the > line... I didn't want to go into the specifics but yes, the LR is noted. > > > Keeping in mind that cost is the most important factor, my searches I've > > found a couple of suggestions - the most promising of which was reading > > the CID from a serial modem. However, I've tried a couple - on of which > > is a BT Enabler that no amount of AT commands can get to give up the CID > > - and concluded that the chances of finding a compatible modem are > > probably slimer than getting the clone to work. > > > Has anyone been able to get the cheap clone cards to offer CID in the > > UK? > > My definition of a cheap clone card is somewhat different from yours, but > for £55.50 + VAT, I use these: > >http://www.voipon.co.uk/openvox-a400p01-p-669.html Given the PSTN rings once a month if that - all the rest is SIP via ITSP, I can't justify the cost of 5 times a telephone for a card - but thanks for the heads up GK old chap. It's not a big issue - but an irritation if anything. I suspect a £2.50 callerID unit, a serial IC and a soldering iron may end up being the solution :-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Audio on pstn call
Try setting the debug level higher, it might give more info to debug .
Core set debug atleast 17
Core set verbose atleast 17
-Irfan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Siti Zalifah Md
Yatim
Sent: Thursday, March 04, 2010 1:51 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] No Audio on pstn call
additional info on the system
Linux home 2.6.30.3-SLACKWARE #1 Sun Feb 7 09:09:33 MYT 2010 i686
Intel(R) Pentium(R) 4 CPU 2.00GHz GenuineIntel GNU/Linux
Asterisk 1.6.2.5, Copyright (C) 1999 - 2009 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
== Parsing '/etc/asterisk/asterisk.conf': == Found
== Parsing '/etc/asterisk/extconfig.conf': == Found
Connected to Asterisk 1.6.2.5 currently running on home (pid = 11838)
Verbosity is at least 7
home*CLI> module show like dahdi
Module Description
Use Count
codec_dahdiGeneric DAHDI Transcoder Codec Translato 0
app_dahdibarge.so Barge in on DAHDI channel application0
chan_dahdi.so DAHDI Telephony Driver 0
app_dahdiscan.so Scan DAHDI channels application 0
app_dahdiras.soDAHDI ISDN Remote Access Server 0
res_timing_dahdi.soDAHDI Timing Interface 0
on the other hand, calls made internally are ok.
On Thu, Mar 4, 2010 at 2:43 PM, Siti Zalifah Md Yatim
wrote:
> Hello,
>
> I'm facing problem where as whenever there are incoming call from
> pstn, there will be no audio coming in. User at the other end also
> could not hear my voice. This happens few days back. Im using asterisk
> 1.6.1.2 with dahdi tool 2.2.0.
>
> I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and
> asterisk 1.6.2.5. However, it does not help at all.
>
> My current config as follows :-
>
> X100P clone card
>
> /etc/dahdi/system.conf
> # Span 1: WCFXO/0 "Generic Clone Board 1" (MASTER)
> fxsks=1
> echocanceller=mg2,1
>
>
> /etc/asterisk/dahdi-channels.conf
> ; Span 1: WCFXO/0 "Generic Clone Board 1" (MASTER)
> ;;; line="1 WCFXO/0/0 FXSKS (SWEC: MG2)"
> signalling=fxs_ls
> callerid=asreceived
> group=0
> context=from-pstn
> channel => 1
> callerid=
> group=
> context=default
>
>
> /etc/asterisk/chan_dahdi.conf
>
> [trunkgroups]
>
>
>
>
> [channels]
> language = my
> ;
> usecallerid = yes
> callwaiting = yes
> usecallingpres = yes
> callwaitingcallerid = yes
> threewaycalling = yes
> transfer = yes
> canpark = yes
> cancallforward = yes
> callreturn = yes
> mailbox = 5000
> echocancel = yes
> echocancelwhenbridged = yes
> rxgain = 2.0
> txgain = 3.0
> group = 1
> callgroup = 1
> pickupgroup = 1
> faxdetect = both
> signalling = fxs_ls
> callerid = asreceived
> group = 0
> channel = 1
> callerid =
> group =
> context = default
> #include "dahdi-channels.conf"
>
>
> my call plan will execute voicemail when there;s incoming call from
> pstn. result as shwon here
>
>
> -- Executing [...@from-pstn:1] Set("DAHDI/1-1", "CallTime=20100304
> 13:45:30") in new stack
> -- Executing [...@from-pstn:2] Set("DAHDI/1-1", "CallerIDString=""
> <01935x>") in new stack
> -- Executing [...@from-pstn:3] System("DAHDI/1-1", "/bin/echo "20100304
> 13:45:30 01935x [] - to pstn" >> /var/log/asterisk/call_log") in
> new stack
> -- Executing [...@from-pstn:4] Answer("DAHDI/1-1", "") in new stack
> -- Executing [...@from-pstn:5] VoiceMail("DAHDI/1-1", "5000,u") in new stack
> -- Stopped music on hold on DAHDI/1-1
> -- Playing 'vm-theperson.gsm' (language 'my')
> -- Playing 'digits/5.gsm' (language 'my')
> -- Playing 'digits/0.gsm' (language 'my')
> -- Playing 'digits/0.gsm' (language 'my')
> -- Playing 'digits/0.gsm' (language 'my')
> -- Playing 'vm-isunavail.gsm' (language 'my')
> -- Playing 'vm-intro.gsm' (language 'my')
> -- Playing 'beep.gsm' (language 'my')
> -- Recording the mes
[asterisk-users] time/date over POTS?
I had a customer ask me about time/date information being sent to his analog (attached to a Linksys SPA2102) answering machine. I didn't know that POTS could carry this information. Is this something Asterisk could send over SIP? Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK CallerID -v- Wildcard W100P
On Thu, 4 Mar 2010, Brian wrote: At the risk of being flamed Has anyone had any success get the 'El cheapo' Wildcard W100P clone's (£20 flavour) to work with UK Caller ID? Looks like there is a patch to Zaptel to make it work: http://www.voip-info.org/wiki/view/Asterisk+and+UK+Caller+ID but who knows if it's in & working... I'm not sure what the status of Asterisk 1.6 is with respect to UK caller ID, being that we have an odd method of sending the FSK ahead of the ring, but I'm guessing I can't be the first to ask this? We do a line polarity reversal, then FSK the caller ID, then ring the line... Keeping in mind that cost is the most important factor, my searches I've found a couple of suggestions - the most promising of which was reading the CID from a serial modem. However, I've tried a couple - on of which is a BT Enabler that no amount of AT commands can get to give up the CID - and concluded that the chances of finding a compatible modem are probably slimer than getting the clone to work. Has anyone been able to get the cheap clone cards to offer CID in the UK? My definition of a cheap clone card is somewhat different from yours, but for £55.50 + VAT, I use these: http://www.voipon.co.uk/openvox-a400p01-p-669.html Gordon-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk & Sofaware & Polycom
*Hello,* * * *Just thought to post our experiences trying to get a Polycom Soundpoint 450 working through a Sofaware to an endpoint doing SIP natting.* * * *As mentioned above our situation was such. We use Asterisk as our PBX and have SIP natted through the corporate firewalls. A remote user has a Polycom 450, and we purchased for him a s...@office 500.* * * *It was a bit of a struggle to get it working, but once we finished it the setup is working like a champ for the user.* * * *The highlight points for anyone attempting anything similar are:* * * *1. If you want to provision the phone using boot options(which I highly suggest), none of the DHCP options in the 500W match option 66 from DHCP. that being said we programmed the Polycom to use a different option. The Avays IP Phone option is 176, so you can configure the phone to use that boot option instead of the default 66. We had to capture the traffic using all three options to find out what they were exactly. Wireshark gave us the exact details needed. Once we knew that you can simply enter the IP of your ftp server used for provisioning.* * * *2. If you are using provisioning like above, definitely look at the NAT options available in the Polycom config files. The latest document I have is: http://www.polycom.com/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf. Check out page A - 151 for the natting options. We ended up not needing the nat.ip option because the sofaware did pretty good natting already. However we used the keepalive, signal and media port options:* * * * * * * *3. The final touch was kind of surprising, the smartdefense options caused more problems, another post on http://sofaware.infopop.cc, mentions disabling both options which worked perfectly, using the console we turned the smart defense option off like so:* * * *set smartdefense ai voip sip alg disable enforce-rfc disabled* * * *It seems that this option turned on caused the connection to time out roughly every 65 seconds. At first this was stumping us as we figured it was a UDP timeout issue on the firewalls, but we dug up the post suggesting to turn it off.* * * *All in all this setup is definitely possible, and seems to work quite well for us. Just thought to post our adventures in case others need to do something similar.* * * *Cheers,* * * *Darrin Henshaw* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UK CallerID -v- Wildcard W100P
At the risk of being flamed Has anyone had any success get the 'El cheapo' Wildcard W100P clone's (£20 flavour) to work with UK Caller ID? I'm not sure what the status of Asterisk 1.6 is with respect to UK caller ID, being that we have an odd method of sending the FSK ahead of the ring, but I'm guessing I can't be the first to ask this? Keeping in mind that cost is the most important factor, my searches I've found a couple of suggestions - the most promising of which was reading the CID from a serial modem. However, I've tried a couple - on of which is a BT Enabler that no amount of AT commands can get to give up the CID - and concluded that the chances of finding a compatible modem are probably slimer than getting the clone to work. Has anyone been able to get the cheap clone cards to offer CID in the UK? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Availstatus returns 20 ?
Hello list.
ChanIsAvail returns 20 for ${AVAILSTATUS}. What does this '20' mean ??
...
exten => 1,n,ChanIsAvail(SIP/sin10)
exten => 1,n,NoOp(chanisavail == ${AVAILSTATUS})
...
[Mar 4 15:10:16] -- Executing [...@sin:7]
ChanIsAvail("IAX2/testlocal-14088", "SIP/sin10") in new stack
[Mar 4 15:10:16] -- Executing [...@sin:8]
NoOp("IAX2/testlocal-14088", "chanisavail == 20") in new stack
What does it mean when ChanIsAvail returns '20' ???
This is what inside /usr/src/asterisk-1.4.25.1/main/devicestate.c :
/*! \brief Device state strings for printing */
static const char *devstatestring[] = {
/* 0 AST_DEVICE_UNKNOWN */ "Unknown", /*!< Valid, but
unknown state */
/* 1 AST_DEVICE_NOT_INUSE */"Not in use", /*!< Not used */
/* 2 AST_DEVICE IN USE */ "In use", /*!< In use */
/* 3 AST_DEVICE_BUSY */ "Busy", /*!< Busy */
/* 4 AST_DEVICE_INVALID */ "Invalid", /*!< Invalid -
not known to Asterisk */
/* 5 AST_DEVICE_UNAVAILABLE */ "Unavailable", /*!< Unavailable
(not registered) */
/* 6 AST_DEVICE_RINGING */ "Ringing", /*!< Ring, ring,
ring */
/* 7 AST_DEVICE_RINGINUSE */"Ring+Inuse", /*!< Ring and in
use */
/* 8 AST_DEVICE_ONHOLD */ "On Hold" /*!< On Hold */
};
Jonas.
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Re: [asterisk-users] Caller ID in Asterisk
Hi Jimmy,
Appreciate your help.
I tried the one below and cudnt get the caller ID.I am getting "Private Call"
and "Out of Area" in the sip phone display when i call from asterisk.
My current extensions.conf looks like below
[general]
static=yes
writeprotect=no
autofallthrough=no
extenpatternmatchnew=no
clearglobalvars=no
priorityjumping=yes
userscontext=default
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=DAHDI/G1
TRUNKMSD=1
[Internal]
include => Incoming
exten => 8001234001,1,Dial(DAHDI/32,,rt)
exten => 8001234002,1,Dial(DAHDI/33,,rt)
exten => 8001234003,1,Dial(DAHDI/34,,rt)
exten => 8001234004,1,Set(CALLERID(num)=8001234004)
exten => 8001234004,n,Set(CALLERID(name)="Line 4")
exten => 8001234004,3,Dial(DAHDI/35,,rt)
exten => 8001234005,1,Dial(DAHDI/36,,rt)
[Incoming]
exten => s,1,Answer
exten => s,2,Dial(DAHDI/g1,20,rt)
exten => _988.,1,Dial(DAHDI/g1/${EXTEN},20)
I also tried changing the dial plan to exten =>
_988.,3,Dial(DAHDI/g1/${EXTEN},20) and in that case the call itself was not
going through
Venugopal
From: asterisk-users-boun...@lists.digium.com on behalf of Jimmy Godbout
Sent: Thu 3/4/2010 5:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID in Asterisk
Hi,
You need to set the callerid before making the call, not after. Also, I guess
it's a typo that the priority in this dialplan is all 1; it should be
exten => 8001234003,1,Set(CALLERID(num)=8001234003)
exten => 8001234003,n,Set(CALLERID(name)="Line 5")
exten => 8001234003,n,Dial(DAHDI/34,,rt)
Unless your using variable for the name and the number, you should not put them
in ${}.
Jimmy
-Original Message-
From: venui...@motorola.com
Sent: Thu, 4 Mar 2010 19:50:03 +0800
To: asterisk-users@lists.digium.com, asterisk-users@lists.digium.com,
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten => 8001234003,1,Dial(DAHDI/34,,rt)
exten => 8001234003,1,Set(CALLERID(num)=${8001234003})
exten => 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
From: asterisk-users-boun...@lists.digium.com on behalf of
Gopalakrishnaiyer Venugopal-Q16770
Sent: Thu 3/4/2010 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
Subject: [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the SIP
server.When i call from the Asterisk machine to the SIP server i am not getting
the caller id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf file
is attached.
Cheers
venu
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Re: [asterisk-users] Caller ID in Asterisk
Hi All,
Please note that this is a lab setup and we are not connected to any external
telcos
Rgds
Venu
From: asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnaiyer
Venugopal-Q16770
Sent: Thu 3/4/2010 5:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users
Mailing List - Non-Commercial Discussion; asterisk-us...@lists.digium.comhi
Subject: Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten => 8001234003,1,Dial(DAHDI/34,,rt)
exten => 8001234003,1,Set(CALLERID(num)=${8001234003})
exten => 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
From: asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnaiyer
Venugopal-Q16770
Sent: Thu 3/4/2010 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
Subject: [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the SIP server.When
i call from the Asterisk machine to the SIP server i am not getting the caller
id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf file is
attached.
Cheers
venu
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Re: [asterisk-users] Best practise for ISDN Video Conferencing..
- "Mark Adams" escreveu: > Hi, thanks for your response. > > I'm not sure if I explained correctly. I need asterisk to provide an > ISDN data function, whilst also routing voice calls over the same PRI. > Is this possible? > > Regards, > Mark > > On 3 Mar 2010, at 17:58, Vinícius Fontes > wrote: > > > - "Mark Adams" escreveu: > > > >> Hi All, > >> > >> I'm about to setup an Asterisk install to take over an old legacy > PBX > >> system. At present, the legacy system has modules in it which > >> provides > >> 4 > >> * data ISDN links to the video conferencing unit (Tandberg 3000 > MXP) > >> on > >> site, these use the ISDN30 (uk) that the normal voice calls go > over. > >> > >> Is it possible to emulate this in asterisk? I've seen zapras but > I'm > >> not > >> sure if that's right. > >> > >> Is there a better way to do Video conferencing over ISDN in > asterisk > >> that will work with the Tandberg unit? > >> > >> Thanks, > >> Mark > >> > > > > I don't think Asterisk can do video over ISDN. It would be great if > > anyone can prove me wrong thought. > > > > -- I understood it perfectly, I guess. You probably have an ISDN videoconferencing product attached to a proprietary PBX (like Siemens, Ericsson, etc). When you make a videoconference call, the Tandberd connects to the PBX, requests X ISDN channels (depends on the video quality you want) and transmits the audio and video as data using the ISDN channels. Unfortunely I'm almost sure Asterisk doesn't supports this. As far as I know, you can only send audio on ISDN channels. There is a VIDEO transfer capability implemented on the CHANNEL function, but I'm not sure if that works at all: pabx:/home/vinicius# asterisk -rx "core show function CHANNEL" -= Info about function 'CHANNEL' =- [Syntax] CHANNEL(item) [Synopsis] Gets/sets various pieces of information about the channel. [Description] Gets/set various pieces of information about the channel. Standard items (provided by all channel technologies) are: (snip) R/W transfercapability ISDN transfer capability (one of SPEECH, DIGITAL, RESTRICTED_DIGITAL, 3K1AUDIO, DIGITAL_W_TONES, or VIDEO). Maybe that could be a start, but unfortunely I have never dealt with a situation like that before. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID in Asterisk
Hi,
You need to set the callerid before making the call, not after. Also, I guess it's a typo that the priority in this dialplan is all 1; it should be
exten => 8001234003,1,Set(CALLERID(num)=8001234003)exten => 8001234003,n,Set(CALLERID(name)="Line 5")
exten => 8001234003,n,Dial(DAHDI/34,,rt)
Unless your using variable for the name and the number, you should not put them in ${}.
Jimmy
-Original Message-From: venui...@motorola.comSent: Thu, 4 Mar 2010 19:50:03 +0800To: asterisk-users@lists.digium.com, asterisk-users@lists.digium.com, asterisk-users@lists.digium.comSubject: Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten => 8001234003,1,Dial(DAHDI/34,,rt)
exten => 8001234003,1,Set(CALLERID(num)=${8001234003})exten => 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
From: asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnaiyer Venugopal-Q16770Sent: Thu 3/4/2010 3:59 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion; asterisk-users@lists.digium.comSubject: [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the SIP server.When i call from the Asterisk machine to the SIP server i am not getting the caller id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf file is attached.
Cheers
venu
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Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten => 8001234003,1,Dial(DAHDI/34,,rt)
exten => 8001234003,1,Set(CALLERID(num)=${8001234003})
exten => 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
From: asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnaiyer
Venugopal-Q16770
Sent: Thu 3/4/2010 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
Subject: [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the SIP server.When
i call from the Asterisk machine to the SIP server i am not getting the caller
id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf file is
attached.
Cheers
venu
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Re: [asterisk-users] Best practise for ISDN Video Conferencing..
Hi, This comes up with a load of results however I can't see anything that relates directly to what I'm talking about. Is anyone doing this in their setup? Cheers, Mark On Thu, Mar 04, 2010 at 02:34:17PM +1300, Alec Davis wrote: > Search bugs.asterisk.org and enter 'digital' in the search field. > > It probably will is my answer. I currently am not using it, so YMMV. > > Alec Davis > > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Adams > Sent: Thursday, 4 March 2010 10:39 a.m. > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Best practise for ISDN Video Conferencing.. > > Hi, thanks for your response. > > I'm not sure if I explained correctly. I need asterisk to provide an ISDN > data function, whilst also routing voice calls over the same PRI. > Is this possible? > > Regards, > Mark > > On 3 Mar 2010, at 17:58, Vinícius Fontes > wrote: > > > - "Mark Adams" escreveu: > > > >> Hi All, > >> > >> I'm about to setup an Asterisk install to take over an old legacy PBX > >> system. At present, the legacy system has modules in it which > >> provides > >> 4 > >> * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) > >> on site, these use the ISDN30 (uk) that the normal voice calls go > >> over. > >> > >> Is it possible to emulate this in asterisk? I've seen zapras but I'm > >> not sure if that's right. > >> > >> Is there a better way to do Video conferencing over ISDN in asterisk > >> that will work with the Tandberg unit? > >> > >> Thanks, > >> Mark > >> > > > > I don't think Asterisk can do video over ISDN. It would be great if > > anyone can prove me wrong thought. > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller ID in Asterisk
Hi All, I have an asterik machine which is connected via a PRI to the SIP server.When i call from the Asterisk machine to the SIP server i am not getting the caller id of the lines at the sip side. Please help me to identify how this can be set.The extensions.conf file is attached. Cheers venu extensions.conf Description: extensions.conf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Identify scripts connecting to the asterisk manager
On Wed, Mar 03, 2010 at 04:41:49PM -0600, Jason Marble wrote: > Is there any easy way to identify which script or service is > connecting to the Asterisk manager? Somewhere on my system a script or > service is trying to connect with a bad user name or password. I get > the following error: connect attempt from '127.0.0.1' unable to > authenticate > > I thought maybe I could do a tcpdump on port 5038 and try to fish out > the bad username or password but I wasn't able to see any passwords or > usernames in plain text. > > Any way I could maybe change the logging in Asterisk to show me the > username that is not able to authenticate? If you can capture the connection attempt in time, use: netstat -ntp -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP / Echo Cancellation
Hello > > I have successfully compiled OSLEC for echo cancellation for DAHDI channel. > > Is there any way to do echo cancellation for SIP Channel. > > Is any, please suggest me.?? > > Thanks in advance.. > > -- > Regards, > > Chandrakant Solanki > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and cellphone/GSM voicemailbox
Hello !
My macro to avoid voicemail of a cellphone is not really working. Can
you take a look at it :
This is the macro :
[macro-testgsm]
exten => s,1,NoOp(inside macro testgsm)
exten => s,n,Wait(2)
exten => s,n,Read(INPUT,,1,1,1)
exten => s,n,GoToIf($["${INPUT}"=="1"]?exit:hangup)
exten => s,n(exit),MacroExit()
exten => s,n(hangup),Hangup()
This is what the CLI shows :
(my zoiper softphone with account testlocal is the caller, sin50 is the
cellphone/GSM)
> [Mar 4 10:36:52] -- SIP/sin50-09f55f80 answered IAX2/testlocal-15506
> [Mar 4 10:36:52] -- Executing [...@macro-testgsm:1]
> NoOp("SIP/sin50-09f55f80", "inside macro testgsm") in new stack
> [Mar 4 10:36:52] -- Executing [...@macro-testgsm:2]
> Wait("SIP/sin50-09f55f80", "2") in new stack
> [Mar 4 10:36:54] -- Executing [...@macro-testgsm:3]
> Read("SIP/sin50-09f55f80", "INPUT||1|1|1") in new stack
> [Mar 4 10:36:54] -- Accepting a maximum of 1 digits.
> [Mar 4 10:37:00] -- User entered nothing.
> [Mar 4 10:37:00] -- Executing [...@macro-testgsm:4]
> GotoIf("SIP/sin50-09f55f80", "0?exit:hangup") in new stack
> [Mar 4 10:37:00] -- Goto (macro-testgsm,s,6)
> [Mar 4 10:37:00] -- Executing [...@macro-testgsm:6]
> Hangup("SIP/sin50-09f55f80", "") in new stack
> [Mar 4 10:37:00] == Spawn extension (macro-testgsm, s, 6) exited non-zero
> on 'SIP/sin50-09f55f80' in macro 'testgsm'
> [Mar 4 10:37:19] == Spawn extension (zoiper, sin, 1) exited non-zero on
> 'IAX2/testlocal-15506'
The CLI says the channel has hung up, but in fact my zoiper softphone is
connected to sin50 and seconds are counting. This way I will surely be
stranded on a voicemail-system !
Can anyone advise ?
Thank you !
Jonas.
On Tue, 2010-03-02 at 14:42 -0500, Fred Posner wrote:
> > Jonas
>
> You can avoid this is several ways... one of the ways I like best is to dial
> with a macro that then requires the recipient to press 1 or some dtmf
> confirmation to accept the call. Very good at avoiding voicemail, cell phone
> service messages, etc.
>
> ---fred
> http://qxork.com
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[asterisk-users] anonymous
hello I would like to implement anonymous calls. is someone who can help me with an idea in extensions.conf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting: Can't fix up channel from 5 to 7 because 7 is already in use, and pri_dchannel: Answer requested on channel 0/7 not in use on span 1
Sometimes I get these in my kernel log: [3763662.549761] __sdla_bus_read_4:888: wanpipe PCI Error: Illegal Register read: 0x0040 = 0x [4014422.079673] __sdla_bus_read_4:888: wanpipe PCI Error: Illegal Register read: 0x0040 = 0x Anyone who knows what this is about? 2010/2/25 Håkon Nessjøen > System have been working great for weeks, using an average 40 of 120 > dahdi channels. > > But today, I suddenly see scary things like this: > >-- Moving call from channel 5 to channel 7 > [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608 > pri_fixup_principle: Can't fix up channel from 5 to 7 because 7 is > already in use > [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:11535 pri_dchannel: > Ringing requested on channel 0/7 not in use on span 1 >-- Moving call from channel 7 to channel 12 > [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608 > pri_fixup_principle: Can't fix up channel from 7 to 12 because 12 is > already in use > [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:11535 pri_dchannel: > Ringing requested on channel 0/12 not in use on span 1 >-- DAHDI/4-1 is ringing > [Feb 25 10:18:17] WARNING[17129]: chan_dahdi.c:10608 > pri_fixup_principle: Can't fix up channel from 5 to 7 because 7 is > already in use > [Feb 25 10:18:17] WARNING[17129]: chan_dahdi.c:11680 pri_dchannel: > Answer requested on channel 0/7 not in use on span 1 > [Feb 25 10:18:22] WARNING[17129]: chan_dahdi.c:10624 > pri_fixup_principle: Whoa, there's no owner, and we're having to fix > up channel 5 to channel 7 > > [Feb 25 10:21:44] WARNING[17129]: chan_dahdi.c:10661 > pri_fixup_principle: Call specified, but not found? > > What would be the reason for things like this to happen? > > And are they really just warnings (as it says), or actual errors, > where something bad is happening to the actual calls, or calls not > acknowledged? > > Håkon > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
