Re: [asterisk-users] Problems Extension with a Call In on Asterisk 1.6
Hi
Very thanks for your helps, that's work very goo
Bye
Olivier
2011/3/25 DHAVAL INDRODIYA :
> Hi Olivier,
>
> here is solutions for your situation , ideally you need to talk with
> Provider and they can set SIP URI
> for given DID numbre , but that can be solved by dial-plan like this.
>
>
> exten => _003318364,1,Set(foo=${SIP_HEADER(To)})
> exten => _003318364,n,Set(cut1=${CUT(foo,:,2)})
> exten => _003318364,n,Set(CLI=${CUT(cut1,>,1)})
> exten => _003318364,n,Set(toexten=${CUT(CLI,@,1)})
> exten => _003318364,n,Noop(ORIGINAL NUMBER : [ ${toexten} ])
> exten => _003318364,n,ExecIf($["${toexten}" =
> "81169"]?Dial(SIP/204,180,rt):Noop(${toexten}))
> exten => _003318364,n,ExecIf($["${EXTEN}" =
> "003318364"]?Dial(SIP/203,180,rt):Noop(${toexten}))
>
>
> On Thu, Mar 24, 2011 at 11:13 AM, Olivier CALVANO
> wrote:
>>
>> Hi
>>
>> Anyone know a solution at my problems ?
>>
>> Thanks
>> Olivier
>>
>>
>>
>>
>>
>>
>>
>> 2011/3/23 Olivier CALVANO :
>> > Hi
>> >
>> > I request your help because i don't have actually a solution at my
>> > problems.
>> >
>> >
>> > I have a Asterisk Server in 1.6
>> > Connected at a SIP Provider
>> > This provider supply me 2 numbers:
>> > 003318364 (official number)
>> > 081169 (Nddi Number)
>> >
>> > When i receive a call on the 081169, he don't use
>> > the extension. He use the 003318364 extension.
>> >
>> > SIP Debug:
>> >
>> > <--- SIP read from UDP://91.121.xxx.xxx:5060 --->
>> > INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0
>> > Allow: UPDATE,REFER,INFO
>> > Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
>> > Contact:
>> > Content-Type: application/sdp
>> > CSeq: 1602837515 INVITE
>> > From: "033426aa"
>> >
>> > ;tag=04459-CI-5fa6f8a1-6f03b5b60
>> > Max-Forwards: 30
>> > P-Preferred-Identity:
>> > To:
>> > User-Agent: Cirpack/v4.42s (gw_sip)
>> > Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2
>> > Content-Length: 481
>> >
>> > v=0
>> > o=cp10 130085910854 130085910854 IN IP4 10.7.1.121
>> > s=SIP Call
>> > c=IN IP4 91.121.bbb.bbb
>> > t=0 0
>> > m=audio 36146 RTP/AVP 18 4 0 8 125 111 101
>> > b=AS:21
>> > a=rtpmap:18 G729/8000/1
>> > a=fmtp:18 annexb=no
>> > a=rtpmap:4 G723/8000/1
>> > a=fmtp:4 annexa=no
>> > a=rtpmap:0 PCMU/8000/1
>> > a=rtpmap:8 PCMA/8000/1
>> > a=rtpmap:125 CLEARMODE/8000/1
>> > a=rtpmap:111 iLBC/8000/1
>> > a=fmtp:111 mode=30
>> > a=rtpmap:101 telephone-event/8000
>> > a=fmtp:101 0-15
>> > a=ptime:30
>> > a=sendrecv
>> > a=sqn:0
>> > a=cdsc: 1 image udptl t38
>> >
>> > <->
>> > --- (13 headers 22 lines) ---
>> > Sending to 91.121.xxx.xxx : 5060 (no NAT)
>> > Using INVITE request as basis request -
>> > 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
>> > Found peer 'Myoperator' for '033426aa' from 91.121.xxx.xxx:5060
>> > Found RTP audio format 18
>> > Found RTP audio format 4
>> > Found RTP audio format 0
>> > Found RTP audio format 8
>> > Found RTP audio format 125
>> > Found RTP audio format 111
>> > Found RTP audio format 101
>> > Peer audio RTP is at port 91.121.bbb.bbb:36146
>> > Found audio description format G729 for ID 18
>> > Found audio description format G723 for ID 4
>> > Found audio description format PCMU for ID 0
>> > Found audio description format PCMA for ID 8
>> > Found unknown media description format CLEARMODE for ID 125
>> > Found audio description format iLBC for ID 111
>> > Found audio description format telephone-event for ID 101
>> > Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d
>> > (g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing),
>> > combined - 0x109 (g723|alaw|g729)
>> > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
>> > (telephone-event), combined - 0x1 (telephone-event)
>> > Peer audio RTP is at port 91.121.bbb.bbb:36146
>> > Looking for 003318364 in Appels-Entrants (domain 78.41.xxx.xxx)
>> >
>> > <--- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 --->
>> > SIP/2.0 404 Not Found
>> > Via: SIP/2.0/UDP
>> > 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx
>> > From: "033426aa"
>> >
>> > ;tag=04459-CI-5fa6f8a1-6f03b5b60
>> > To: ;tag=as50e04b6a
>> > Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
>> > CSeq: 1602837515 INVITE
>> > Server: Asterisk PBX 1.6.1.8
>> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>> > Supported: replaces, timer
>> > Content-Length: 0
>> >
>> >
>> > <>
>> > [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527
>> > handle_request_invite: Call from '0033459aa' to extension
>> > '003318364' rejected because extension not found.
>> > Scheduling destruction of SIP dialog
>> > '04459-nk-5fa6f8a0-18641f...@sip.myoperator.net' in 6400 ms (Method:
>> > INVITE)
>> > <--- SIP read from UDP://91.121.xxx.xxx:5060 --->
>> > ACK sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0
>> > Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myop
Re: [asterisk-users] Removing Polycom Transfer Softkey
From the polycom pdf: divert.fwd.x.enabled If set to 1, the user will be able to enable universal call forwarding through the soft key menu. This sounds like it turns on and turns off the call forwarding feature on the phone. I can try it out Monday, but I don't see where it has any relation to transfer (both attended and blind). On 03/27/2011 08:43 PM, C F wrote: In phone.cfg set the following line to divert.fwd.1.enabled="0" from: divert.fwd.1.enabled="1" For more info check page 323: http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf On Fri, Mar 25, 2011 at 6:33 PM, Mark Murawski wrote: Sorry for the crosspost. This was supposed to be on -users I know some of you are polycom gurus... Anyone know how to remove transfer from a polycom 33x phone? We've set allowtransfer=no, but we would like to remove a polycom soft key as well. -- -- _ -- 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] Removing Polycom Transfer Softkey
In phone.cfg set the following line to divert.fwd.1.enabled="0" from: divert.fwd.1.enabled="1" For more info check page 323: http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf On Fri, Mar 25, 2011 at 6:33 PM, Mark Murawski wrote: > Sorry for the crosspost. This was supposed to be on -users > > > I know some of you are polycom gurus... > > Anyone know how to remove transfer from a polycom 33x phone? We've set > allowtransfer=no, but we would like to remove a polycom soft key as well. > > -- > _ > -- 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] CDR MYSQL missing field data
On Sunday 27 March 2011 19:36:45 Eric W. Davenport wrote: > Hello, > > I have Asterisk-1.8.3.2, dahdi-linux-complete-2.4.1+2.4.1, and > libpri-1.4.11.5 installed and running on a Ubuntu 10.04 server all built > from source. > > Everything is working nicely except one small issue. > > The CDR records are stored in the CSV file correctly and complete. > > The MySQL storage is working as it should and is automatically updating > all the fields except the CLID field. > > I have compared and constructed and destructed the system 3 times since > Thursday and I cannot figure out why the field does not get populated. > > If I run the import from csv script it correctly populates the CLID so I > believe that tables are setup correct. > > Has any one seen this and could possibly point me at the offending conf > file. > > I am more familiar than I want to be with cdr_.conf files and I > cannot find where the problem is. > > I have browsed all the wiki's, blogs, and emails looking for a hint and > I did not find anything. > > Anything would be appreciated. Do you have "alias callerid => clid" in your cdr_mysql.conf file (in the [columns] context)? -- Tilghman -- _ -- 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] CDR MYSQL missing field data
Hello, I have Asterisk-1.8.3.2, dahdi-linux-complete-2.4.1+2.4.1, and libpri-1.4.11.5 installed and running on a Ubuntu 10.04 server all built from source. Everything is working nicely except one small issue. The CDR records are stored in the CSV file correctly and complete. The MySQL storage is working as it should and is automatically updating all the fields except the CLID field. I have compared and constructed and destructed the system 3 times since Thursday and I cannot figure out why the field does not get populated. If I run the import from csv script it correctly populates the CLID so I believe that tables are setup correct. Has any one seen this and could possibly point me at the offending conf file. I am more familiar than I want to be with cdr_.conf files and I cannot find where the problem is. I have browsed all the wiki's, blogs, and emails looking for a hint and I did not find anything. Anything would be appreciated. Thank you, Eric -- Eric W. Davenport Cert-In Software Systems, Inc. P.O. Box 346 Bakersville, NC 28705 800-873-0110 ewdavenp...@certin.com www.certin.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] spa8000 spa2102 t38 faxing
On 28/03/2011 5:48 AM, Israel Gottlieb wrote: still no luck i hear it change to t38 but it just doesnt connect Do you have two fax devices at your end, even a fax-modem attached to a computer will do? You are going to need to provide more information such as your current configuration and traces of the sessions. If you turn off all T.38 options in Asterisk and on the SPA you should still be able to make a transmissions using the G711 codecs. Can you confirm you are able to send a facsimile from your device using a PSTN line? Larry. -- _ -- 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] spa8000 spa2102 t38 faxing
still no luck i hear it change to t38 but it just doesnt connect On Sun, Mar 27, 2011 at 5:26 AM, Larry Moore wrote: > Perhaps this will help. > > I have a SPA8800 which has 4 x FXS & 4 x FXO ports. > > It took me some time to produce a working configuration. > > In Asterisk I have the following where 904 is the extension of the > fax-modem and itsp is you VoIP Service Provider. > > sip.conf > > [general] > . > . > faxdetect=no > t38pt_udptl=yes,redundancy,maxdatagram=400 > . > . > > [904] > ; Cisco SPA8800 FXS Port 4 > ; Analogue FAX Modem attached > type=friend > defaultuser=904 > secret= > call-limit=2 > qualify=yes > canreinvite=no > directmedia=no > directrtpsetup=no > ignoresdpversion=yes > transport=udp,tcp > host=dynamic > context= > faxdetect=no > > . > . > [itsp] > . > . > faxdetect=yes > ignoresdpversion=yes > . > . > > > I am including information from my SPA8800 for one of the FXS ports I have > a Fax Modem attached to, the key to getting it to work I believe is the "FAX > Tone Detect Mode". > > Audio Configuration > > Preferred Codec: G711a Second Preferred Codec: Unspecified > Third Preferred Codec: UnspecifiedUse Pref Codec Only: no > Silence Supp Enable: yes Silence Threshold: medium > G729a Enable: no Echo Canc Enable: yes > G723 Enable: no Echo Canc Adapt Enable: yes > G726-16 Enable: no Echo Supp Enable: yes > G726-24 Enable: no FAX CED Detect Enable: yes > G726-32 Enable: no FAX CNG Detect Enable: yes > G726-40 Enable: no FAX Passthru Codec: G711a > DTMF Process INFO: yes FAX Codec Symmetric: yes > DTMF Process AVT: yes FAX Passthru Method: ReINVITE > DTMF Tx Method: AVT DTMF Tx Mode: Strict > DTMF Tx Strict Hold Off Time: 40FAX Process NSE: no > Hook Flash Tx Method: None FAX Disable ECAN: no > Release Unused Codec: yes FAX Enable T38: yes > FAX T38 Redundancy: 1 FAX Tone Detect Mode: callee only > Symmetric RTP: yes > > Supplementary Service Settings > > CW Setting: noBlock CID Setting: no > Block ANC Setting: noDND Setting: no > CID Setting: yes CWCID Setting: yes > Dist Ring Setting: yesSecure Call Setting: no > Message Waiting: noAccept Media Loopback Request: automatic > Media Loopback Mode: sourceMedia Loopback Type: media > > Larry. > > -- > _ > -- 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] pbx.c: We were unable to say the number
Oh crap, you're right, my bad. Yes, I also agree, it's most probably the
language and/or missing files
On Sun, Mar 27, 2011 at 4:30 PM, Jeff LaCoursiere wrote:
> On Sun, 2011-03-27 at 16:14 -0500, Sherwood McGowan wrote:
> >
> >
> > On Sun, Mar 27, 2011 at 2:50 PM, Mohammad Khan
> > wrote:
> > Here is the dialplan in macro:
> >
> > exten => s,n,SayNumber($[${ARG1} % 100])
> >
> > when 662 was passed as ARG1, I had the following at log:
> >
> > WARNING[15217] pbx.c: We were unable to say the number 62, is
> > it too large?
> >
> > Do you see any odd in my dialplan?
> >
> >
> >
> >
> > 662 % 100 = 66.2, not 62. It seems to me that there's more going on
> > here..Maybe Asterisk is being confused by actually getting 66.2? I'm
> > not readily able to look into the source, but I think that Asterisk
> > (or at least, SayNumber) cannot handle a number with a decimal point,
> > but please don't take that as gospel.
> >
>
> '%' is 'modulus', and 62 is the correct result. I am betting it is the
> language setting, and missing audio files.
>
> j
>
>
>
> --
> _
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> http://www.asterisk.org/hello
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] pbx.c: We were unable to say the number
On Sun, 2011-03-27 at 16:14 -0500, Sherwood McGowan wrote:
>
>
> On Sun, Mar 27, 2011 at 2:50 PM, Mohammad Khan
> wrote:
> Here is the dialplan in macro:
>
> exten => s,n,SayNumber($[${ARG1} % 100])
>
> when 662 was passed as ARG1, I had the following at log:
>
> WARNING[15217] pbx.c: We were unable to say the number 62, is
> it too large?
>
> Do you see any odd in my dialplan?
>
>
>
>
> 662 % 100 = 66.2, not 62. It seems to me that there's more going on
> here..Maybe Asterisk is being confused by actually getting 66.2? I'm
> not readily able to look into the source, but I think that Asterisk
> (or at least, SayNumber) cannot handle a number with a decimal point,
> but please don't take that as gospel.
>
'%' is 'modulus', and 62 is the correct result. I am betting it is the
language setting, and missing audio files.
j
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Re: [asterisk-users] pbx.c: We were unable to say the number
On Sun, Mar 27, 2011 at 2:50 PM, Mohammad Khan wrote:
> Here is the dialplan in macro:
>
> exten => s,n,SayNumber($[${ARG1} % 100])
>
> when 662 was passed as ARG1, I had the following at log:
>
> WARNING[15217] pbx.c: We were unable to say the number 62, is it too large?
>
> Do you see any odd in my dialplan?
>
>
> 662 % 100 = 66.2, not 62. It seems to me that there's more going on
here..Maybe Asterisk is being confused by actually getting 66.2? I'm not
readily able to look into the source, but I *think* that Asterisk (or at
least, SayNumber) cannot handle a number with a decimal point, but please
don't take that as gospel.
If Tilghman's question doesn't result in a fix, the next thing I'd say is to
check and make sure that a floating point number can be supplied as an
argument to SayNumber.
Also make sure you're logging verbose, debug, error, and warning messages
into a logfile, bump the verbosity and debug up to 5, and then run another
test call that will result in the number(s) you're testing. Next, send off a
larger amount of the content from the logfile, the single WARNING line is
not enough for anything more than a blind guess. For instance, it would be
REALLY fantastic is you would send the log line that displays SayNumber
actually being executed, like ( *[DATE&TIME] VERBOSE[23609] pbx.c: --
Executing [s@contextname:priority] SayNumber("CHANNELNAME", "66.2") in new
stack *)... It would be even MORE fantastic if you included almost ALL of
that call's log output, but at the very LEAST there should be around 5
lines, starting from the verbose output for execution of SayNumber. That
way, we don't just get the warning message you're complaining about, but the
EXACT executions and messages outputted leading up to the warning message.
But then again, you could continue to do essentially the same thing and hope
for different results...
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Re: [asterisk-users] pbx.c: We were unable to say the number
On Sunday 27 March 2011 14:50:37 Mohammad Khan wrote:
> Here is the dialplan in macro:
>
> exten => s,n,SayNumber($[${ARG1} % 100])
>
> when 662 was passed as ARG1, I had the following at log:
>
> WARNING[15217] pbx.c: We were unable to say the number 62, is it too
> large?
>
> Do you see any odd in my dialplan?
What do you have CHANNEL(language) set to at the time? What language
packs do you have installed? What is the exact version of Asterisk you
have installed?
Usually, what this error indicates is that you have one or more sound files
missing, unreadable, or in a format that cannot be transcoded to the codec
you're using.
--
Tilghman
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[asterisk-users] DAHDI custom ring cadences in 1.8.3
When did Dial() with a custom ring cadence replace the default from indications.conf for subsequent calls? indications.conf: ringcadence = 2000,4000 asterisk -rx "dahdi show cadences" r1: 667,1333 extensions.conf: exten=> 201, 1, Dial(DAHDI/1) exten=> 202, 1, Dial(DAHDI/1r1) exten=> 203, 1, Dial(DAHDI/1r0) Dialing, in sequence: 201 -> 2000,4000; Good 202 -> 667,1333 ; Good 201 -> 667,1333 ; Huh? 203 -> 2000,4000; r0 seems to be default cadence 201 -> 2000,4000 Do I now need to specify r0 every time I dial an FXS line to be sure it will use the default cadence in case I've previously used a custom one? -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pbx.c: We were unable to say the number
Here is the dialplan in macro:
exten => s,n,SayNumber($[${ARG1} % 100])
when 662 was passed as ARG1, I had the following at log:
WARNING[15217] pbx.c: We were unable to say the number 62, is it too large?
Do you see any odd in my dialplan?
On Sat, Mar 26, 2011 at 2:44 PM, Sherwood McGowan <
sherwood.mcgo...@gmail.com> wrote:
> Again, the relevant dialplan code is important. It is quite possible that
> there's an issue with the dialplan code that you (as the person who's
> dealing with the issue) may have missed. It happens all the time.
>
>
>
>
> On Sat, Mar 26, 2011 at 1:25 PM, Mohammad Khan wrote:
>
>> I am using asterisk 1.4.38
>> I am getting this warning occasionally when executing SayNumber in a macro
>> with argument which is less than 100.
>>
>>
>> On Sat, Mar 26, 2011 at 11:03 AM, Sherwood McGowan <
>> sherwood.mcgo...@gmail.com> wrote:
>>
>>>
>>>
>>> On Fri, Mar 25, 2011 at 11:15 PM, Mohammad Khan wrote:
>>>
Hello,
Occasionally, I get the following warning in my asterisk log,
pbx.c: We were unable to say the number [n], is it too large?
n is two or one digit number, which doesn't look like large to me!
Could anybody please tell more about this warning, like in what scenario
I may have this warning.
Thanks,
Mohammad
>>>
>>> Please post the relevant context that is being executed, that'll give us
>>> not only the actual application, but more info as to how it's being passed.
>>>
>>>
>>
>>
>
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Re: [asterisk-users] Asterisk 1.8 Packages for Debian and Ubuntu
> This effort is not intended to replace packaging of Asterisk in the > official Debian or Ubuntu repositories. Our repositories are for > providing access to major versions of Asterisk that are newer than what > is included. We are exploring ways to work as closely as possible with > the Debian and Ubuntu package maintainers to ensure that we do not > duplicate efforts and that we provide the best possible result for users > of Asterisk. Thanks for providing these - can you just clarify your policy on the following: - file locations - same layout as the regular Debian packages? - upgrade policy - is it intended that someone who has Debian 6 with the existing Asterisk 1.6 packages (from Debian's maintainer) can just upgrade to the Digium package without moving or changing any config? -- _ -- 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] [SOLVED] Back-to-back asterisk PRI issue
After following changes my D-Channel comes up and its working!!! :) vi /etc/wanpipe/wanpipe*.conf TDMV_DCHAN = 0 TDMV_HWEC = NO @Thanks all of them who helped here... No beer for others ;) -S From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 23:44:31 + Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue Check out this https://issues.asterisk.org/view.php?id=17270 > From: tilgh...@meg.abyt.es > To: asterisk-users@lists.digium.com > Date: Fri, 25 Mar 2011 17:23:28 -0500 > Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue > > On Friday 25 March 2011 16:23:27 satish patel wrote: > > I just start "Pri set debug on span 1" and its showing D-channel is > > down > > How do you have the underlying T1 signalling set up in > /etc/dahdi/system.conf (on both ends)? > > -- > Tilghman > > -- > _ > -- 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] Jabber/Jingle to Google users via local XMPP server
Hi all, All the examples I've come across seem to suggest configuring jabber.conf/jingle.conf/gtalk.conf for a real Google account. What about the scenario where the Asterisk server should connect to an account on a private Jabber server and using Jingle (voice calling over Jabber)? e.g. for the domain widgets.com: - there is a copy of ejabberd running on the same box as Asterisk, and Asterisk registers to it using the jabber ID aster...@widgets.com - DNS is configured so that u...@widgets.com can chat to u...@gmail.com (already working, testing with a chat client such as Empathy or Psi) Google user frie...@gmail.com wants to make a voice call to aster...@widgets.com - is it possible? For this scenario, is gtalk.conf needed at all? Is gtalk.conf needed for any Jabber server, such as the ejabbard instance described above? Regards, Daniel -- _ -- 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] Streaming Hold Music
One minor comment: On Sun, Mar 27, 2011 at 02:41:11AM -0400, Alexander Lopez wrote: > In musiconhold.conf > [default] > mode=custom > directory=/var/lib/asterisk/mohmp3-empty If you don't use 'mode=files', there's no need to set 'directory'. > application=/etc/asterisk/bin/mohstream.sh [snip] -- 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
