[asterisk-users] sip show channels shows non-existent channels on 1.6.0.19 and 1.4.27.1 ?
I never do sip show channels but i tried it this morning to see if everything is working after upgrading 2 boxes to 1.6.0.19 and 1.4.27.1 Is it correct that Asterisk doesn't clean up sip channels anymore after using them ? On one box i can see sip lines for every phone that was attempted to call, on the other box i see sip channels for 2 sip peers i register too. This is the output on the console: 1.4.27.1 : pbx*CLI> sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 1.1.1.1 mysipname 576687fc4a8 00573/0 0x0 (nothing) No 10.0.0.0 221 774a1331284 00102/0 0x0 (nothing) No Init: NOTIFY 10.0.0.1 224 0359f0d23a4 00102/0 0x0 (nothing) No Init: NOTIFY etc. 1.6.0.19 : Peer User/ANRCall ID Format Hold Last Message 8.0.0.0 (None) d34db33f-125982 0x0 (nothing)No Rx: OPTIONS 8.0.0.0 (None) d34db33f-125982 0x0 (nothing)No Rx: OPTIONS 2 active SIP dialogs Ip addressed modified (nuked) for the above examples Is this normal? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flashing Cisco 7941 to SIP
You need to purchase a smartnet license for the phone in question in order to legally get the sip firmware. Thanks, --Warren Selby On Dec 2, 2009, at 11:28 PM, "Ricardo Melendez" wrote: > Hi to All, I am trying to flash to SIP image one Cisco 7941 IP Phone > to work with Asterisk, I have searched the internet and find some > instructions but I need the firmware SIP Image to complete the flash. > > > > Can anyone help me with the SIP image for Cisco 7941? > > > > The image name is > > > > cmterm-7941_7961-sip.8-5-2.cop > > > > > > > > Thanks in advance. > > > > Ricardo > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feature Request: "SayNumberFiles"
Why not do something with Background()? i.e
Playback(you-have)
SayNumber(${numMessages})
Playback(messages)
Background(press-1-or-2)
Then just be sure to record the audio files in the appropriate
directory...
Thanks,
--Warren Selby
On Dec 3, 2009, at 12:39 AM, Olivier wrote:
> Hi,
>
> Currently, it seems impossible to use the output of SayNumber
> application as an input to Read application.
> So, if you want to develop an IVR with something like "You've got 23
> messages. Type 1 to listen to the first one. Type 2 to leave", you
> must parse this message into 3 pieces and want for the last one play
> to start listening of user input :
> Playback ( "You've got")
> SayNumber (23)
> Read ("messages. Type 1 to listen to the first one. Type 2 to
> leave", ...)
>
> Being to get the list of sound files that SayNumber would play would
> be very convenient to build concatenate several one into one
> temporary file. When played with Read app, user wouldn't have to
> wait for the last part to type its answer :
>
> myfilelist = SayNumberFiles(23)
> tempfile=concat( "You've got", myfilelist, "messages. Type 1 to
> listen to the first one. Type 2 to leave")
> Read (tempfile, ...)
>
> Did I miss a workaround ?
> What would you say about this feature request ?
>
> Regards
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Featuremap help
Using version 1.2.35 built by root @ slate on a i686 running Linux on 2009-09-15 00:24:10 UTC Problem - I cannot get featuremap right. Have added a feature that I want to direct to an extension in extension.conf Extension is 521 In features.conf - [applicationmap] dumpcaller => #9,callee,goto(521|1) show features - Dynamic Feature Default Current --- --- --- dumpcallerno def #9 Result -- Feature Found: dumpcaller exten: dumpcaller Dec 1 20:08:02 WARNING[18659]: res_features.c:958 feature_exec_app: Could not find application (goto(521|1)) I have tried many variations per docs. It shows goto(521|1) also tried (521,1) and dial(local/521/n) and many others. Always get could not find application. Any ideas? Doug ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Feature Request: "SayNumberFiles"
Hi,
Currently, it seems impossible to use the output of SayNumber application as
an input to Read application.
So, if you want to develop an IVR with something like "You've got 23
messages. Type 1 to listen to the first one. Type 2 to leave", you must
parse this message into 3 pieces and want for the last one play to start
listening of user input :
Playback ( "You've got")
SayNumber (23)
Read ("messages. Type 1 to listen to the first one. Type 2 to leave", ...)
Being to get the list of sound files that SayNumber would play would be very
convenient to build concatenate several one into one temporary file. When
played with Read app, user wouldn't have to wait for the last part to type
its answer :
myfilelist = SayNumberFiles(23)
tempfile=concat( "You've got", myfilelist, "messages. Type 1 to listen to
the first one. Type 2 to leave")
Read (tempfile, ...)
Did I miss a workaround ?
What would you say about this feature request ?
Regards
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[asterisk-users] Flashing Cisco 7941 to SIP
Hi to All, I am trying to flash to SIP image one Cisco 7941 IP Phone to work with Asterisk, I have searched the internet and find some instructions but I need the firmware SIP Image to complete the flash. Can anyone help me with the SIP image for Cisco 7941? The image name is cmterm-7941_7961-sip.8-5-2.cop Thanks in advance. Ricardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi_tool shows no alarms, but no line connected
Hi, I'm using Sangoma's wanpipe together with dahdi, all software downloaded today at most recent version. Hardware is Sangoma A104, a 4xE1 card. Installation went well. Anyway, wanrouter status shows a different result than dahdi_tool or dahdi_scan. I've just put a hardware loop on port 1. All the other ports are open. wanrouter status shows the expected result: Device name | Protocol | Station | Status| wanpipe1| AFT TE1 | N/A | Connected | wanpipe2| AFT TE1 | N/A | Disconnected | wanpipe3| AFT TE1 | N/A | Disconnected | wanpipe4| AFT TE1 | N/A | Disconnected | However: # dahdi_scan 2 [2] active=yes alarms=OK description=wanpipe2 card 1 name=WPE1/1 manufacturer= devicetype= location= basechan=1 totchans=31 irq=0 type=digital- syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS Why are the dahdi tools not reflecting the values by wanrouter? Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable Name needed
Thank you that was it
James Shigley
Monroe Telephone Answering Service
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: Wednesday, December 02, 2009 4:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Variable Name needed
My question is, Does anyone know what variable I would use to get the
information for "To" from these SIP calls, the below is the actual SIP
packet obtained from the CLI with SIP Debug On. Other than I stripped
out the IPs
The variable you are seeking is ${SIP_HEADER(TO)}
I parse the SIP headers from callcentric like this:
Set(calldest=${CUT(CUT(SIP_HEADER(To),@,1),:,2)})
Which gives me a real US number like 1xx.
Credit for the parsing syntax goes to someone else (not sure where I
found it online).
--Dave
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Re: [asterisk-users] Problem with Timeout
Thanks for your replies. Am I correct that if I use "session-timers=refuse", asterisk will never disconnect a call? That could be quite expensive if a call gets lost. Any idea why the line I am using in the dialplan isn't working? Thanks Dan Dan Journo wrote: > > Hi, > > I have a problem with incoming calls. They all seem to be ending after > 600 seconds (10 minutes). > > I've added:- > > exten => _X.,2,Set(TIMEOUT(absolute)=18000) > > However the calls seem to still be ending after 600 seconds. > > I've checked the debug and verbose and the line above is being executed. > > Is there any way to increase the timeout globally? Using sip.conf or > something similar? > > Thanks > > Dan > > > > ___ > -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskNOW sip module not installed
Hey Guys, I have installed AsteriskNOW and I've found that the SIP module is not installed (along with all the SIP configuration files) until I log into the freepbx webGUI and click the reload button. Is there a way to get all the sip configuration files and module installed without having to do a reload from the webGUI? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable Name needed
My question is, Does anyone know what variable I would use to get the
information for "To" from these SIP calls, the below is the actual SIP packet
obtained from the CLI with SIP Debug On. Other than I stripped out the IPs
The variable you are seeking is ${SIP_HEADER(TO)}
I parse the SIP headers from callcentric like this:
Set(calldest=${CUT(CUT(SIP_HEADER(To),@,1),:,2)})
Which gives me a real US number like 1xx.
Credit for the parsing syntax goes to someone else (not sure where I found it
online).
--Dave
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[asterisk-users] Rsrvd state and off hook dahdi issue
Hello all, I have asterisk 1.4.26.2 working on linux CentOS 2.6.18 with a dahdi driver 2.2.0.2 and Digium Wildcard AEX410. I configured 3 ports of 4 all fxs. The port 1 is connected to a analogue phone, the port 3 and 4 are connected to two analogue faxes. dahdi-channels.conf: ;;; line="1 WCTDM/0/0 FXOKS" threewaycalling=yes transfer=yes signalling=fxo_ks callerid="John Doe" <1234> callgroup=1 pickupgroup=1 context=national_mobile channel => 1 ;;; line="3 WCTDM/0/2 FXOKS" threewaycalling=no transfer=no signalling=fxo_ks callerid=FAX Three callgroup= pickupgroup= context=national_landline channel => 3 ;;; line="4 WCTDM/0/3 FXOKS" threewaycalling=no transfer=no signalling=fxo_ks callerid=FAX RH callgroup= pickupgroup= context=national_landline channel => 4 the system.conf file: fxoks=1 echocanceller=mg2,1 fxoks=2 echocanceller=mg2,2 fxoks=3 echocanceller=mg2,3 fxoks=4 echocanceller=mg2,4 # Global data loadzone= pt defaultzone = pt Some times the the fxs lines stays off hook permanently, with the state reserved. I tried to do a soft hangup with no results. The only thing that works is to restart the dahdi drivers. Anyone has an idea of how to resolve this?? Thanks in advance, Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: Variable Name needed
It might be worth mentioning the voip call is coming from a number we
have thru bandwidth.com in case anyone uses them.
James Shigley
From: James A. Shigley
Sent: Wednesday, December 02, 2009 3:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Variable Name needed
That wasn't it either. I tried a few other likely fields from that page
none of which gave the correct data
James Shigley
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Wednesday, December 02, 2009 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Variable Name needed
According to this link
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List I'd
go with ${SIPCALLID}
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Wednesday, December 02, 2009 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable Name needed
Other than having stripping out IPs this is what I am receiving for my
voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works
fine calls that come in on my PRI. BUT at least from this VOIP source
the To field which is my RDNIS information for these calls, doesn't
actually fill into ${CALLERID(rdnis). But as you can see I'm getting the
information.
My question is, Does anyone know what variable I would use to get the
information for "To" from these SIP calls, the below is the actual SIP
packet obtained from the CLI with SIP Debug On. Other than I stripped
out the IPs
<--- Transmitting (no NAT) to:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP;branch=z9hG4bK6631.98b11517.0;received=
Via: SIP/2.0/UDP;branch=z9hG4bK6631.f18e9a97.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK506071629460-1256675031807
Record-Route:
Record-Route:
From: "BEAUMONT TX"
;tag=VPSF506071629460
To: ;tag=as4b59d217
Call-ID: DALMGC0520091202194656056692@
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact:
Content-Length: 0
<>
Thank You for your time, and I apologize if this is a repeat question. I
did Google, and search thru my * email archive (back thru April 09) for
an answer first.
James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments.
<>___
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Re: [asterisk-users] Variable Name needed
That wasn't it either. I tried a few other likely fields from that page
none of which gave the correct data
James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Wednesday, December 02, 2009 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Variable Name needed
According to this link
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List I'd
go with ${SIPCALLID}
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Wednesday, December 02, 2009 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable Name needed
Other than having stripping out IPs this is what I am receiving for my
voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works
fine calls that come in on my PRI. BUT at least from this VOIP source
the To field which is my RDNIS information for these calls, doesn't
actually fill into ${CALLERID(rdnis). But as you can see I'm getting the
information.
My question is, Does anyone know what variable I would use to get the
information for "To" from these SIP calls, the below is the actual SIP
packet obtained from the CLI with SIP Debug On. Other than I stripped
out the IPs
<--- Transmitting (no NAT) to:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP;branch=z9hG4bK6631.98b11517.0;received=
Via: SIP/2.0/UDP;branch=z9hG4bK6631.f18e9a97.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK506071629460-1256675031807
Record-Route:
Record-Route:
From: "BEAUMONT TX"
;tag=VPSF506071629460
To: ;tag=as4b59d217
Call-ID: DALMGC0520091202194656056692@
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact:
Content-Length: 0
<>
Thank You for your time, and I apologize if this is a repeat question. I
did Google, and search thru my * email archive (back thru April 09) for
an answer first.
James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments.
<>___
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Re: [asterisk-users] b option in Directory
On Wed, 2009-12-02 at 09:40 -0600, Danny Nicholas wrote: > There are indeed lots of improvements, but if the OP NEEDS some features > that automagically worked in Zaptel that still don't in DAHDI (POTS Line > supervision...) *Please* don't continue to bring up this example in the mailing lists. >From everything I've been able to find in my research, there is *no difference* in answer supervision (or far-end disconnect supervision, for that matter) between Zaptel and DAHDI. I've explained this once before, but I'll explain it yet again so that hopefully Google will index it for the next person that comes along and asks it: Let's assume I'm making an outbound call on an analog phone line connected to an FXO port in my Asterisk system. In the United States, the telephone companies don't send any kind of signaling to let me know that the far end has answered my call. Hence the reason Asterisk treats *all* outbound calls on FXO ports as having been answered, unless you go changing settings in zapata.conf/chan_dahdi.conf (like setting "answeronpolarityswitch=yes"). Obviously telephone signaling can and does vary from country to country, which is why have settings like "answeronpolarityswitch" and "hanguponpolariyswitch". In short, there's no difference in Zaptel and DAHDI in this regard, so please don't keep using it as an excuse for people to stick with Zaptel. If in fact there were any regressions of this nature in the transition from Zaptel to DAHDI, rest assured that we would have corrected them by now. -- Jared Smith Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable Name needed
According to this link
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List I'd go
with ${SIPCALLID}
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Wednesday, December 02, 2009 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable Name needed
Other than having stripping out IPs this is what I am receiving for my voip
calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works fine calls
that come in on my PRI. BUT at least from this VOIP source the To field
which is my RDNIS information for these calls, doesn't actually fill into
${CALLERID(rdnis). But as you can see I'm getting the information.
My question is, Does anyone know what variable I would use to get the
information for "To" from these SIP calls, the below is the actual SIP
packet obtained from the CLI with SIP Debug On. Other than I stripped out
the IPs
<--- Transmitting (no NAT) to:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP;branch=z9hG4bK6631.98b11517.0;received=
Via: SIP/2.0/UDP;branch=z9hG4bK6631.f18e9a97.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK506071629460-1256675031807
Record-Route:
Record-Route:
From: "BEAUMONT TX" ;tag=VPSF506071629460
To: ;tag=as4b59d217
Call-ID: DALMGC0520091202194656056692@
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact:
Content-Length: 0
<>
Thank You for your time, and I apologize if this is a repeat question. I did
Google, and search thru my * email archive (back thru April 09) for an
answer first.
James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If you
are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information is prohibited. If
you have received this email in error, please notify the sender immediately
by "reply to sender only" message and destroy all electronic and hard copies
of the communication, including attachments.
cid:image003.png@01C9F268.65A4F5C0
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[asterisk-users] Variable Name needed
Other than having stripping out IPs this is what I am receiving for my
voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works
fine calls that come in on my PRI. BUT at least from this VOIP source
the To field which is my RDNIS information for these calls, doesn't
actually fill into ${CALLERID(rdnis). But as you can see I'm getting the
information.
My question is, Does anyone know what variable I would use to get the
information for "To" from these SIP calls, the below is the actual SIP
packet obtained from the CLI with SIP Debug On. Other than I stripped
out the IPs
<--- Transmitting (no NAT) to:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP;branch=z9hG4bK6631.98b11517.0;received=
Via: SIP/2.0/UDP;branch=z9hG4bK6631.f18e9a97.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK506071629460-1256675031807
Record-Route:
Record-Route:
From: "BEAUMONT TX"
;tag=VPSF506071629460
To: ;tag=as4b59d217
Call-ID: DALMGC0520091202194656056692@
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact:
Content-Length: 0
<>
Thank You for your time, and I apologize if this is a repeat question. I
did Google, and search thru my * email archive (back thru April 09) for
an answer first.
James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
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Re: [asterisk-users] Help configuring Audiocodes MP-104 FXO
I've set at Protocol Management >> FXO Settings >> Dialing Mode ==> One
Stage and everything is fine now
Thank you very much John,
EDA
On Wed, Dec 2, 2009 at 1:43 PM, John Balogh wrote:
> > I want to do single-stage dialing. I've just realized I
>
> > have the two-stage running now (I get dial tone and then,
>
> > when i introduce the number, the call get through).
>
>
>
> Right.
>
>
>
> According to the SIP User's Manual
>
> LTRT-65405 MediaPack SIP User's Manual Ver 4.6.pdf
>
> page 67/294
>
>
>
> "
>
> Enable Digit Delivery to Tel [EnableDigitDelivery]
>
> Disable [0] = Disabled (default).
>
> Enable [1] = Enable Digit Delivery feature for MediaPack/FXO & FXS.
>
> The digit delivery feature enables sending of DTMF digits to the gateway’s
> port after the line is offhooked (FXS) or seized (FXO). For IP->Tel calls,
> after the line is offhooked / seized, the MediaPack plays the DTMF digits
> (of the called number) towards the phone line.
>
> [...]
>
> To use this feature with FXO gateways, configure the gateway to work in one
>
> stage dialing mode.
>
> "
>
>
>
> You probably need to set the above.
>
>
>
> The FXO parameter (from page 107/294):
>
>
>
> "
>
> Dialing Mode [IsTwoStageDial]
>
> One Stage [0] = One-stage dialing.
>
> Two Stage [1] = Two-stage dialing (default).
>
> Used for IP->FXO gateways calls.
>
>
>
> If two-stage dialing is enabled, then the FXO gateway seizes one of the
> PSTN/PBX lines without performing any dial, the remote user is connected
> over IP to PSTN/PBX, and all further signaling (dialing and Call Progress
> Tones) is performed directly with the PBX without the gateway’s
> intervention.
>
>
>
> If one-stage dialing is enabled, then the FXO gateway seizes one of the
> available lines (according to Channel Select Mode parameter), and dials the
> destination phone number received in INVITE message. Use the ‘Waiting For
> Dial Tone’ parameter to specify whether the dialing should come after
> detection of dial tone, or immediately after seizing of the line.
>
> "
>
>
>
> So you probably need to clear that parameter (it is not configured in your
> .INI file now, so you need to add it, or change the web interface drop-down
> control).
>
>
>
> Let us know if this helps.
>
>
>
> JDB
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Daniel - Asterisk
>
> *Sent:* Wednesday, December 02, 2009 12:33 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Help configuring Audiocodes MP-104 FXO
>
>
>
> Hi list,
>
>
> I'm trying to get ready the MP-104 FXO to use qith my box, but when I send
> calls I hear only dial tone and after a few seconds I get busy signal.
>
> I very appreciate your advices.
>
> Command line results and SIPconfigurations follows:
>
> *CLI>*
> -- Executing [7991696...@total:1] Playback("SIP/101-09dd8918", "beep")
> in new stack
> -- Playing 'beep' (language 'es')
> -- Executing [7991696...@total:4] Dial("SIP/101-09dd8918",
> "SIP/201/991696900") in new stack
> -- Called 201/991696900
> -- SIP/201-09ddc890 answered SIP/101-09dd8918
>
>
> *sip.conf*
> [201]
> secret =
> callerid = Mobile_01 <201>
> type = friend
> host = dynamic
> context = total
> dtmfmode=rfc2833
> qualify = yes
> call-limit=5
> disallow = all
> allow = gsm
> allow = ulaw
> allow = alaw
> allow = g729
>
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Help configuring Audiocodes MP-104 FXO
> I want to do single-stage dialing. I've just realized I
> have the two-stage running now (I get dial tone and then,
> when i introduce the number, the call get through).
Right.
According to the SIP User's Manual
LTRT-65405 MediaPack SIP User's Manual Ver 4.6.pdf
page 67/294
"
Enable Digit Delivery to Tel [EnableDigitDelivery]
Disable [0] = Disabled (default).
Enable [1] = Enable Digit Delivery feature for MediaPack/FXO & FXS.
The digit delivery feature enables sending of DTMF digits to the
gateway's port after the line is offhooked (FXS) or seized (FXO). For
IP->Tel calls, after the line is offhooked / seized, the MediaPack plays
the DTMF digits (of the called number) towards the phone line.
[...]
To use this feature with FXO gateways, configure the gateway to work in
one
stage dialing mode.
"
You probably need to set the above.
The FXO parameter (from page 107/294):
"
Dialing Mode [IsTwoStageDial]
One Stage [0] = One-stage dialing.
Two Stage [1] = Two-stage dialing (default).
Used for IP->FXO gateways calls.
If two-stage dialing is enabled, then the FXO gateway seizes one of the
PSTN/PBX lines without performing any dial, the remote user is connected
over IP to PSTN/PBX, and all further signaling (dialing and Call
Progress Tones) is performed directly with the PBX without the gateway's
intervention.
If one-stage dialing is enabled, then the FXO gateway seizes one of the
available lines (according to Channel Select Mode parameter), and dials
the destination phone number received in INVITE message. Use the
'Waiting For Dial Tone' parameter to specify whether the dialing should
come after detection of dial tone, or immediately after seizing of the
line.
"
So you probably need to clear that parameter (it is not configured in
your .INI file now, so you need to add it, or change the web interface
drop-down control).
Let us know if this helps.
JDB
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel -
Asterisk
Sent: Wednesday, December 02, 2009 12:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Help configuring Audiocodes MP-104 FXO
Hi list,
I'm trying to get ready the MP-104 FXO to use qith my box, but when I
send calls I hear only dial tone and after a few seconds I get busy
signal.
I very appreciate your advices.
Command line results and SIPconfigurations follows:
CLI>
-- Executing [7991696...@total:1] Playback("SIP/101-09dd8918",
"beep") in new stack
-- Playing 'beep' (language 'es')
-- Executing [7991696...@total:4] Dial("SIP/101-09dd8918",
"SIP/201/991696900") in new stack
-- Called 201/991696900
-- SIP/201-09ddc890 answered SIP/101-09dd8918
sip.conf
[201]
secret =
callerid = Mobile_01 <201>
type = friend
host = dynamic
context = total
dtmfmode=rfc2833
qualify = yes
call-limit=5
disallow = all
allow = gsm
allow = ulaw
allow = alaw
allow = g729
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[asterisk-users] Asterisk-Addons 1.4.10, 1.6.0.4, and 1.6.1.2 Now Available
The Asterisk Development Team has announced several Asterisk-Addons releases, including Asterisk-Addons 1.4.10, 1.6.0.4, and 1.6.1.2. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve several community reported issues, primarily in the format_mp3 and res_config_mysql modules. * Fix audio problems with format_mp3. (closes issue #15850. Reported and Tested by 99gixxer. Patched by russell.) * Fix memory corruption caused by format_mp3. (closes several issues. Reported by jvandal, axisinternet, aragon, maxnuv, amorson, jensvb, marhbere, thom4fun. Tested by the same plus zerohalo, rgj, russell.) * Fix MP3 problem with Playback() and Background(). (closes issue #15432. Reported and Tested by DLNoah. Patched by dvossel.) * Select uncommented lines, not commented ones (res_config_mysql). (closes issue #15746. Reported by makoto. Patched by Tilghman.) For a full list of changes in these releases, please see the ChangeLogs: http://svn.asterisk.org/svn/asterisk-addons/tags/1.4.10/ChangeLog http://svn.asterisk.org/svn/asterisk-addons/tags/1.6.0.4/ChangeLog http://svn.asterisk.org/svn/asterisk-addons/tags/1.6.1.2/ChangeLog Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialout problem with analog phone
Hi,
I am stuck with my analog telephones on DADHI/1.
I can place a call from a SIP-phone to the analog phone.
But as soon as I try to dial somethin on the analog phone asterisk does hang-
up immediately.
What do I miss?
Cheers
Eckhard
My extensions.conf
[default]
exten => 1001,1,Answer()
exten => 1001,2,Playback(hello-world)
exten => 1001,3,Hangup()
exten => 10,1,Dial(DAHDI/1)
exten => s,1,Dial(SIP/${EXTEN})
[sipphones]
exten => 8010,1,Dial(SIP/8010,20)
exten => 8010,2,VoiceMail(8010,u)
exten => 8011,1,Dial(SIP/8011,20)
exten => 8011,2,VoiceMail(8011,u)
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Re: [asterisk-users] Help configuring Audiocodes MP-104 FXO
Please post your BOARD.INI file (configuration of AudioCodes).
Also, do you expect to do single-stage dialing (MP104 takes SIP invite
information and turns that into DTMF output), or two-stage dialing
(MP104 only answers and connects the audio/RTP path)?
John Balogh, Sr. Systems Engineer
PSU, ITS, TNS, Network Planning
sip:j...@psu.edu
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel -
Asterisk
Sent: Wednesday, December 02, 2009 12:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Help configuring Audiocodes MP-104 FXO
Hi list,
I'm trying to get ready the MP-104 FXO to use qith my box, but when I
send calls I hear only dial tone and after a few seconds I get busy
signal.
I very appreciate your advices.
Command line results and SIPconfigurations follows:
CLI>
-- Executing [7991696...@total:1] Playback("SIP/101-09dd8918",
"beep") in new stack
-- Playing 'beep' (language 'es')
-- Executing [7991696...@total:4] Dial("SIP/101-09dd8918",
"SIP/201/991696900") in new stack
-- Called 201/991696900
-- SIP/201-09ddc890 answered SIP/101-09dd8918
sip.conf
[201]
secret =
callerid = Mobile_01 <201>
type = friend
host = dynamic
context = total
dtmfmode=rfc2833
qualify = yes
call-limit=5
disallow = all
allow = gsm
allow = ulaw
allow = alaw
allow = g729
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[asterisk-users] Help configuring Audiocodes MP-104 FXO
Hi list,
I'm trying to get ready the MP-104 FXO to use qith my box, but when I send
calls I hear only dial tone and after a few seconds I get busy signal.
I very appreciate your advices.
Command line results and SIPconfigurations follows:
*CLI>*
-- Executing [7991696...@total:1] Playback("SIP/101-09dd8918", "beep")
in new stack
-- Playing 'beep' (language 'es')
-- Executing [7991696...@total:4] Dial("SIP/101-09dd8918",
"SIP/201/991696900") in new stack
-- Called 201/991696900
-- SIP/201-09ddc890 answered SIP/101-09dd8918
*sip.conf*
[201]
secret =
callerid = Mobile_01 <201>
type = friend
host = dynamic
context = total
dtmfmode=rfc2833
qualify = yes
call-limit=5
disallow = all
allow = gsm
allow = ulaw
allow = alaw
allow = g729
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[asterisk-users] Asterisk Queues Tutorial updated - Hot-Desking without Agent Channels
Hello, Just to let you know, our popular tutorial on setting up Asterisk for call centres has been updated. The tutorial covers everything from initial Asterisk installation to full call centre configuration with dynamic login and hot-desking support for agents. The old version of the tutorial used Agent Channels (e.g. Agent/1001) to distribute calls to agents through the Queue() application. These have been deprecated since Asterisk 1.4, and the AgentCallbackLogin function is not supported as of Asterisk 1.6 The new version of our tutorial shows you how to achieve the same functionality (and more) with Local Channels, which are the recommended replacement for Agent Channels. So, if you are setting up Asterisk for call centre use, or if you are currently using Agent Channels and want to maintain your upgrade path, please take a look at http://www.orderlyq.com/asteriskqueues.html and find out how it's done. Thanks for reading. Matt King, CEO Orderly Software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] b option in Directory
There are indeed lots of improvements, but if the OP NEEDS some features that automagically worked in Zaptel that still don't in DAHDI (POTS Line supervision...) Enough digressing... Here's the patch link https://issues.asterisk.org/view.php?id=7151 According to dates on the page, it should apply to 1.4.21 with no problems. It's only 16 lines of code if you had to apply it manually. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrick Hartman Sent: Wednesday, December 02, 2009 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] b option in Directory On 12/02/2009 08:32 AM, Martin Roy wrote: > I'm running asterisk 1.4.21 and I see if I go on the wiki that there's a b option that let you enter the first name OR last name of a user. I see that to make this work I need a patch. I'm wondering how can I install this patch as it's an option one of my customer would like to have but I never had to deal with patch before. I usually just take the release version of asterisk and install it as is. > > P.S. I would like to keep the version 1.4.21 because it's the last version that I know of that use Zaptel by default instead of DAHDI. You do know that you should be able to compile against Zaptel throughout the 1.4.x series. It's worth the effort to upgrade to Dahdi though. Several improvements. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT - Oreka Call Recording
- Original Message - From: "Alex Balashov" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, December 1, 2009 5:09:51 PM Subject: Re: [asterisk-users] Slightly OT - Oreka Call Recording It's easy enough to build Oreka from source that trying to solve the dependency issues is not worth it. After doing much searching on the issues I tend to agree with you. I of course was looking for an easier way... :-) --Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT - Oreka Call Recording
- Original Message - From: "David Backeberg" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, December 2, 2009 9:04:23 AM Subject: Re: [asterisk-users] Slightly OT - Oreka Call Recording On Wed, Dec 2, 2009 at 6:08 AM, Leif Neland wrote: > /me wonders why you don't let asterisk record the audio itself instead of > adding a 3.rd party There was a discussion about this a few months ago on this list. I haven't personally run into this issue, but another poster claimed that at a certain point of simultaneous channels, they were having problems writing to disk and that load was going high on the system. When they separated recording out, those problems went away. At the time, the idea sounded silly, and the more I thought about it, SIP audio over UDP is the perfect candidate for this kind of port-mirroring treatment. I have not played with Oreka, but after that previous discussion it sounds interesting enough that I want to play with it in my lab. Also, what if you need a call recording solution in a hosted environment. Would you rather setup one system to transparently record all calls flowing through your network or install some sort of call recording mechanism on many dozens of machines? And, dare I say, what if you needed to perform call recording in a covert fashion when you did not have access to the system itself? --Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] b option in Directory
On 12/02/2009 08:32 AM, Martin Roy wrote: > I'm running asterisk 1.4.21 and I see if I go on the wiki that there's a b > option that let you enter the first name OR last name of a user. I see that > to make this work I need a patch. I'm wondering how can I install this patch > as it's an option one of my customer would like to have but I never had to > deal with patch before. I usually just take the release version of asterisk > and install it as is. > > P.S. I would like to keep the version 1.4.21 because it's the last version > that I know of that use Zaptel by default instead of DAHDI. You do know that you should be able to compile against Zaptel throughout the 1.4.x series. It's worth the effort to upgrade to Dahdi though. Several improvements. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT - Oreka Call Recording
On Wed, Dec 2, 2009 at 6:08 AM, Leif Neland wrote: > /me wonders why you don't let asterisk record the audio itself instead of > adding a 3.rd party There was a discussion about this a few months ago on this list. I haven't personally run into this issue, but another poster claimed that at a certain point of simultaneous channels, they were having problems writing to disk and that load was going high on the system. When they separated recording out, those problems went away. At the time, the idea sounded silly, and the more I thought about it, SIP audio over UDP is the perfect candidate for this kind of port-mirroring treatment. I have not played with Oreka, but after that previous discussion it sounds interesting enough that I want to play with it in my lab. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fsk callerid with DTAS start(like dtmf in issue 9096)?
hi: our country use ETSI Standard ETS 300 659-1 to send caller-id. the caller-id format may be DTMF or FSK. with the newest patch in issue 9096 (9096-1.6.2-branch.diff), the DTMF part is solved now. but asterisk still can not detect the FSK part. in the standard the FSK will send with a DT-AS (dual tone alerting signal) start. the method is described in pdf below(page 12): http://www.araxinfo.com/~bacvic/ets_30065901.pdf I wonder if asterisk can treat this kind of FSK just like the DTMF in issue 9096. issue 9096 add option "cidstart=dtmf" to chan_dahdi.conf. can we add "cidstart=dtas" or "cidstart=fsk" to detect FSK also? is there a easy way to patch FSK like the DTMF? I may need to spent time to study the patch. I hope someone can give me some advice or direction. thanks a lot for help!! Regards, tbskyd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] b option in Directory
Martin Roy wrote: > I'm running asterisk 1.4.21 and I see if I go on the wiki that there's a b > option that let you enter the first name OR last name of a user. I see that > to make this work I need a patch. I'm wondering how can I install this patch > as it's an option one of my customer would like to have but I never had to > deal with patch before. I usually just take the > I guess I should slow down when I read, it would appear that you already have the patch. Copy it into the Asterisk source directory and type: patch -p0 < name.of.patch And then re-compile asterisk and do a make install. Copy the sound file to the /var/lib/asterisk/sounds directory. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] b option in Directory
Martin Roy wrote: > I'm running asterisk 1.4.21 and I see if I go on the wiki that there's a b > option that let you enter the first name OR last name of a user. I see that > to make this work I need a patch. I'm wondering how can I install this patch > as it's an option one of my customer would like to have but I never had to > deal with patch before. I usually just take the > I've got the patch and the necessary sound file if you'd like me to send it to you off list. It applies to 1.4.20.1 fine, you'll need to see if it will apply to your version. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Timeout
FWIW, session-timers is also a valid option in the 1.4 tree (lot's of 1.6 things aren't). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Wednesday, December 02, 2009 8:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with Timeout Hi I friend had a similar problem and he sorted it out with the following in the sip.conf (this is in 1.6) session-timers=refuse Ish Dan Journo wrote: > > Hi, > > I have a problem with incoming calls. They all seem to be ending after > 600 seconds (10 minutes). > > I've added:- > > exten => _X.,2,Set(TIMEOUT(absolute)=18000) > > However the calls seem to still be ending after 600 seconds. > > I've checked the debug and verbose and the line above is being executed. > > Is there any way to increase the timeout globally? Using sip.conf or > something similar? > > Thanks > > Dan > > > > ___ > -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] b option in Directory
I'm running asterisk 1.4.21 and I see if I go on the wiki that there's a b option that let you enter the first name OR last name of a user. I see that to make this work I need a patch. I'm wondering how can I install this patch as it's an option one of my customer would like to have but I never had to deal with patch before. I usually just take the release version of asterisk and install it as is. P.S. I would like to keep the version 1.4.21 because it's the last version that I know of that use Zaptel by default instead of DAHDI. Thanks Martin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Timeout
Hi I friend had a similar problem and he sorted it out with the following in the sip.conf (this is in 1.6) session-timers=refuse Ish Dan Journo wrote: > > Hi, > > I have a problem with incoming calls. They all seem to be ending after > 600 seconds (10 minutes). > > I’ve added:- > > exten => _X.,2,Set(TIMEOUT(absolute)=18000) > > However the calls seem to still be ending after 600 seconds. > > I’ve checked the debug and verbose and the line above is being executed. > > Is there any way to increase the timeout globally? Using sip.conf or > something similar? > > Thanks > > Dan > > > > ___ > -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect Two Asterisk's using isdn Cards
mos...@infolog.mr escribió: It's 2 T1/E1 cards! Specifically, on of it is a TE110P and the other is a TE122! Hi, That would be a very special need, I'm wondering why connect two asterisk with expensive E1/T1 cards when you can connect them with simple network cards and use SIP or IAX2? Anyway, the way to do it is to define one asterisk as the master (network side) and the other one as the slave (CPE side). You can achieve that configuring one box with signalling = pri_net and the other one with signalling = pri_cpe in chan_dahdi.conf. Thank you for the answer i tried this, and it works well ( Two servers are synchronized). Now i want to make a call between the servers. I have a sip phone at each end. How would i configure my asterisk files to get it? Chapters 4 and 5 of the TFOT Second Edition book will help you get going with this: http://downloads.oreilly.com/books/9780596510480.pdf The only thing is to configure the appropriate extensions to Dial() through the DAHDI trunk between servers. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
Hi Philipp! Philipp Kempgen schrieb: > Where exactly did you register your DNS server? Did your registrar > handle it for you? http://www.nic.at ? http://www.enum.at ? > Yes, my registrar http://www.my-enum.at handles it. (my-enum.at seems to be a sub-company of nic.at) First you have to register your telephone number (which is done after validating that it's really your number). Then you can *either* enter service-uri's to that number (sip: mailto: ) *or* "assign" a "domain specific Nameserver" to that nuber. Norbert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about g729
Alex Balashov wrote: > My understanding is that Asterisk will not pass through calls in codecs > for which it does not have support and/or licenses; it simply does not > advertise them in the SDP negotiation. 'support' - yes, 'licenses' - no. Asterisk supports passthrough, recording and playback of quite a few codecs for which there are no transcoding modules available at all (G.723.1, H.263/4, etc). G.729 falls into this category if there is no transcoding module loaded, or if there are no licenses available. The only time that Asterisk will not offer G.729 in an outbound negotiation is if the incoming channel is not in G.729 (so thus would require transcoding) and there is no transcoding path available. The same is true for other codecs that don't have transcoding available... but by definition, this is not 'passthrough'. Also, to clarify an earlier point, Digium makes all the standard Asterisk prompt sets available in G.729 format, so the built-in applications can be used on G.729 channels without requiring transcoding. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Timeout
Hi, I have a problem with incoming calls. They all seem to be ending after 600 seconds (10 minutes). I've added:- exten => _X.,2,Set(TIMEOUT(absolute)=18000) However the calls seem to still be ending after 600 seconds. I've checked the debug and verbose and the line above is being executed. Is there any way to increase the timeout globally? Using sip.conf or something similar? Thanks Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
Norbert Zawodsky schrieb: > BTW, meantime I have alread implemented all that. My DNS server is up & > running. > I've chosen one of the existing registrars and payed him for registering > 7.6.5.4.3.2.1.1.3.4.e164.arpa as "my" number at nic.at. > The registrar vaildated that this really is my number, and when this was > confirmed, did the registration. > Then I registerd my DNS server as the authorative master for the domain > *.7.6.5.4.3.2.1.1.3.4.e164.arpa Where exactly did you register your DNS server? Did your registrar handle it for you? http://www.nic.at ? http://www.enum.at ? > That's it. It works! That's good news! > When ever anyone anywhere in the world does a ENUMLOOKUP(mynumber), my > server receives a request and (hopefully) sends the correct answer. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect Two Asterisk's using isdn Cards
>>> >> It's 2 T1/E1 cards! >> Specifically, on of it is a TE110P and the other is a TE122! >> >> > Hi, > > That would be a very special need, I'm wondering why connect two > asterisk with expensive E1/T1 cards when you can connect them with > simple network cards and use SIP or IAX2? > > Anyway, the way to do it is to define one asterisk as the master > (network side) and the other one as the slave (CPE side). You can > achieve that configuring one box with signalling = pri_net and the other > one with signalling = pri_cpe in chan_dahdi.conf. Thank you for the answer i tried this, and it works well ( Two servers are synchronized). Now i want to make a call between the servers. I have a sip phone at each end. How would i configure my asterisk files to get it? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT - Oreka Call Recording
- Original Message - From: Tim Nelson To: asterisk-users@lists.digium.com Sent: Wednesday, December 02, 2009 12:06 AM Subject: [asterisk-users] Slightly OT - Oreka Call Recording Greetings all- I'd like to install Oreka on a Centos 5.x server for monitoring my Asterisk systems(using port mirroring) but I find I'm having problems with the version of libpcap installed. /me wonders why you don't let asterisk record the audio itself instead of adding a 3.rd party Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about g729
My understanding is that Asterisk will not pass through calls in codecs for which it does not have support and/or licenses; it simply does not advertise them in the SDP negotiation. Perhaps I am wrong. Dan Journo wrote: > However, I've read somewhere that passthrough doesnt require a license. Which > means that if your sip clients can transmit in g729 and your voip provider > can receive in g729, your asterisk server won't need to do any encoding and > therefore doesn't need any licenses. It is simply passing the data through > from your sip clients to the voip provider. Not sure what happens if you want > to play recorded messages and things. It would probably need licenses then > because its encoding. > > > Sent from my Windows Mobile® phone. > > -Original Message- > From: Alex Balashov > Sent: 02 December 2009 01:13 > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Question about g729 > > > All calls. > > Landy Landy wrote: > >>> You only need to purchase 10 licenses, if all 10 clients >>> will be making calls at the same time. >> Ok. Does this apply only for outbound calls using a voip provider and/or >> applies to calls within the lan? >> >> >> >> >> >> ___ >> -- 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 > > > -- > Alex Balashov - Principal > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Featuremap help
Doug Crompton wrote: > Using version 1.2.35 built by root @ slate on a i686 running Linux on > 2009-09-15 00:24:10 UTC > > Problem - I cannot get featuremap right. > > Have added a feature that I want to direct to an extension in > extension.conf > > > I'm not 100% sure, but I don't think that option was available until the 1.4.x series. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about g729
Sorry for the repetition. I didn't see the other responses. -Original Message- From: Thomas Kenyon Sent: 02 December 2009 07:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about g729 Tilghman Lesher wrote: > On Tuesday 01 December 2009 19:10:08 Alex Balashov wrote: >> All calls. >> >> Landy Landy wrote: You only need to purchase 10 licenses, if all 10 clients will be making calls at the same time. >>> Ok. Does this apply only for outbound calls using a voip provider and/or >>> applies to calls within the lan? > > An additional clarification: it only applies to calls in which codecs need to > be transcoded. If you have a g729 call bridged to another g729 call, then no > license is used in that call path. > Also, the only consideration, isn't the endpoints. If the call is being recorded or you are in a conference, then the call needs to be transcoded for mixing purposes. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about g729
However, I've read somewhere that passthrough doesnt require a license. Which means that if your sip clients can transmit in g729 and your voip provider can receive in g729, your asterisk server won't need to do any encoding and therefore doesn't need any licenses. It is simply passing the data through from your sip clients to the voip provider. Not sure what happens if you want to play recorded messages and things. It would probably need licenses then because its encoding. Sent from my Windows Mobile® phone. -Original Message- From: Alex Balashov Sent: 02 December 2009 01:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about g729 All calls. Landy Landy wrote: >> You only need to purchase 10 licenses, if all 10 clients >> will be making calls at the same time. > > Ok. Does this apply only for outbound calls using a voip provider and/or > applies to calls within the lan? > > > > > > ___ > -- 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
