[asterisk-users] Mexican ITSP needed
Hey all, I was wondering if anyone knows about a Mexican ITSP I can connect to to route calls from and to my * boxen. If it matters: I'm located in The Netherlands and one of our customers is in Mexico so if we need a Mexican presence that is not an issue. Thanks. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] DEVICE_STATE() and Asterisk 1.6.0.10
Why are you putting semi-colons at the end of every line? The dialplan
isn't written in PHP ;).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L.
Kline
Sent: 15 July 2009 23:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mark Michelson wrote:
You need to set a call-limit for the SIP peer. Device state
calculation for a
SIP peer is predicated on both the call-limit and busylevel. Let's say
that you
were to have a call-limit of 2, but no busylevel set. These are the
device
states reported for the peer based on the number of calls currently
handled:
Hi Mark. Thanks for your explanation of these parameters.
I should have posted my configurations. I double-checked the contents
of sip.conf and I have this. The 'subscribecontext' was added for
testing, per the other reply I got for my question.
;
; Settings common to all devices on our system
;
[basic-options](!)
type=friend
host=dynamic
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
qualify=yes
;
; Standard desksets here
;
[lan-deskset](!,basic-options)
context=sip-deskset
notifyringing = yes
notifyhold = yes
limitonpeers = yes
call-limit=99
[6668](lan-deskset)
secret=mysecret
callerid=Matts SIP 6668
username=Barry's IP450
call-limit=32
busylevel=1
subscribecontext=hint-context
My hint-context is:
[hint-context]
exten = 6668,hint,SIP/6668;
I'm still not getting anything other than NOT_INUSE from DEVICE_STATE.
Here is the CLI output:
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:1]
NoOp(SIP/-0955ecc8, SIP/6668 has state NOT_INUSE) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:2]
NoOp(SIP/-0955ecc8, SIP/ has state NOT_INUSE) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:3]
ExecIf(SIP/-0955ecc8, 0?Busy(10)) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:4]
Dial(SIP/-0955ecc8, SIP/6668) in new stack
And here is sip show inuse:
corp-asterisk*CLI sip show inuse
* User name In use Limit
6668 1 32
6667 0 99
1 99
* Peer name In use Limit
6668 1/1/0 32
6667 0/0/0 99
0/0/0 99
For completeness, here is the dialplan that's producing this:
exten = 6668,1,NoOp(SIP/${EXTEN} has state
${DEVICE_STATE(SIP/${EXTEN})});
exten = 6668,n,NoOp(SIP/ has state ${DEVICE_STATE(SIP/)});
exten =
6668,n,ExecIf($[${DEVICE_STATE(SIP/${EXTEN})}=BUSY]?Busy(10));
exten = 6668,n,Dial(SIP/${EXTEN});
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)
iD8DBQFKXlwaCFu3bIiwtTARAkRpAJ4+2WF9qrIwrC3Kdpwd0YAOm/5S1wCfUR1T
CtI9kZNQYpW2Sv6uFNud7Jo=
=9Zp/
-END PGP SIGNATURE-
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[asterisk-users] Sending things to Jabber but not within an extension
I have set my Asterisk server up to connect to my Jabber server and send messages with the caller ID details in them to the recipients of incoming calls - this is working very nicely. There are a few other things I can think of right now that I would like to send to Jabber but as yet I do not know whether they are possible. They are: (a) a count of messages in a voicemail box - triggered when the user connects to Jabber. (b) a notification that a voicemail was left - triggered when the caller leaving the message hangs up. (c) a notification that a specific SIP peer has become unreachable - or better still, has been unreachable for five minutes. As I say, I do not know if any of these things are possible, nor how to do them if they are. I have looked at what is on voip-info.org and what the book has but have not seen anything that seems relevant. Of course, I may be right that these are not possible at present, or I may have missed or misunderstood how to do it. I am on version 1.4.21.2~dfsg-1-pmr-2 - which is from Debian stable plus a locally added patch. Thanks in advance to anyone who can advise, even if just to confirm that it is not possible yet. -- Phil Reynolds mail: phil-aster...@tinsleyviaduct.com Web: http://www.tinsleyviaduct.com/phil/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 This message was sent using IMP, the Internet Messaging Program. ___ -- 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] Generic question about PBX PRI installs
Jerry Geis wrote: The PBX guy seems to always complain about how he has MANY options and thats not enough information... What else am I supposed to supply this person. Are they not the PBX expert?... Anyway as example. the last customer I told the above information. He set up the PBX and I can make 4 digit calls successfully, 7 digit and long distance are not successful. They are hitting some error condition that the call is going to the switch board. So the connection is working just not completely working. This is connecting to a nortel 1000 pbx. I dont know anything about a nortel switch to them what to change. What should I be supplying to these PBX guys to get the installs going smoother and quicker? I have connected my Nortel Meridian to * via PRI successfully(?). Nortel has what is called BARS/NARS. The ability to dial numbers to access certain routes is based on the Network Class of Service (NCOS) of a telephone or trunk. I would suspect that the user has the trunk set with a NCOS to low to access the outbound routes. Also, tandem calls from the Tie line to the outside trunks will need to dial the proper access code to activate BARS/NARS. (Ie. If a Nortel user needs to dial 9, so does the call from the tie trunk.) There is also the option, instead of using the BARS/NARS access code, the tie can use the route access code (ACOD) directly. As far as the PRI setup, I have found that I prefer QSIG (if the Nortel system release supports it) because if will properly pass all the CLID info between systems, NI2 did not. I hope this helps. Dale No virus found in this outgoing message. Checked by AVG - www.avg.com Version: 8.5.387 / Virus Database: 270.13.16/2241 - Release Date: 07/16/09 05:58:00 ___ -- 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] AGI to announce temperature from weather.com XML file
I would like to have the ability to have Asterisk announce the temperature -- not using TTS -- within the dialplan. For a non-Asterisk project, I have a cron job that periodically pulls down an XML file from weather.com containing local weather data (TWC's user agreement requires that data be cached locally). Using sed, I also create a text file that contains only the numeric value of the current temperature, created from that XML file (e.g. tmp65/tmp in the XML file becomes a text file with 65 as its only contents). I am hoping someone on the list has an example of a lightweight AGI script that I may modify to either read the simple text file and set a dialplan variable to the current temperature, or hopefully a more-sophisticated one which will parse the XML file to set the dialplan variable. The end goal is to have Asterisk play the speech files temperature sixty five degrees or the equivalent non-English files per the channel's current language setting. Thank you. Any assistance will be greatly appreciated. Sincerely, Trevor Hammonds ___ -- 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] Struggling with Macros and s Extension
Hi all,
I'm sure this has been done before but I just can't figure it out.
On my * box I have a simple IVR:
[tolc_menu] ; Welcome and information to callers
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Background(welcome-to-tolc) ; Say Hello
exten = s,n,Wait(1)
exten = s,n(tryagain),Background(enter-ext-of-personor) ; Enter
extension number if known, or
exten = s,n,Background(pls-stay-on-line) ; Trying to connect...
exten = s,n,WaitExten(5)
exten = s,n,Macro(belllord,${ALANL}${ALANB},303)
exten = _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
exten = _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
.
.
.
Hopefully you'll see that the caller can either enter an extension
number or wait. If they wait, we use macro-belllord:
[macro-belllord]
exten = s,1,Dial(${ARG1},20,t)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten = s-NOANSWER,n,Hangup()
exten = s-BUSY,1,Voicemail(${ar...@business,b)
exten = s-BUSY,n,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1)
The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202
If I call in, dial the extension (say 101) and connect, then the Link
Event on the AMI port (and in CDRs) correctly displays *both* numbers of
the connection.
For that scenario we use macro-call_extention:
[macro-call_extension]
exten = s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten = s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
exten = s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
exten = s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
exten = _s-.,1,Goto(s-NOANSWER,1)
If however I wait and let macro-belllord do it's stuff. I only ever see
s as the called party's number.
I really need to know what that extension number is.
Could someone help me and show how I can rejig this? It was suggested to
do something with ${MACRO_EXTEN} but I can't get it at all...
Many thanks in advance.
Alan
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Re: [asterisk-users] advices on how to debridge/rebridge a call?
Sebastian Maz wrote: this is what I'm trying to accomplish: - receiving an inbound call from A - dialing another number (B) - bridge A and B - every x minutes, debridge A and B, and bridge A with C (SIP call to an platform that is gonna play an ad) - rebridge A and B Any advice on how to do this? Dial cmd with the use of the M or G option? Asterisk Manager? AGI? Thanks for your ideas There is the Bridge() application in Asterisk 1.6.x that may be useful. You could probably use some sort of script that connects to the manager, and uses that application to move the bridges around. Leif Madsen http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to enable dial to a...@asterisk.blurb.com
John A. Sullivan III wrote: On Wed, 2009-07-15 at 14:34 +1000, Alex Samad wrote: The subject line says it all how do I enable this style of call. Pointers to the dns setup and asterisk setup would be great snip If I understand what you are seeking, you can try these URIs: http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial http://www.blyon.com/blog/index.php/2009/06/22/p2p-sip-uri-dialing/ snip Great posts (especially the blyon.com one). If you're looking for a DNS provider (assuming you run your own domain name, but don't have your own DNS servers), then I've found www.editdns.net is great for this kind of thing. They allow you to setup SRV records (unlike some of the other free DNS hosters I've used). Note that I'm not affiliated with editdns.net in any way -- just found them when trying to do the same thing :) Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI to announce temperature from weather.com XMLfile
I have just the thing in PHP. Drop me a personal e-mail and I'll whiz it over. Andrew Thomas Technical Services Manager a...@datavox.co.uk DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Trevor Hammonds Sent: 16 July 2009 12:50 To: 'Asterisk Users List' Subject: [asterisk-users] AGI to announce temperature from weather.com XMLfile I would like to have the ability to have Asterisk announce the temperature -- not using TTS -- within the dialplan. For a non-Asterisk project, I have a cron job that periodically pulls down an XML file from weather.com containing local weather data (TWC's user agreement requires that data be cached locally). Using sed, I also create a text file that contains only the numeric value of the current temperature, created from that XML file (e.g. tmp65/tmp in the XML file becomes a text file with 65 as its only contents). I am hoping someone on the list has an example of a lightweight AGI script that I may modify to either read the simple text file and set a dialplan variable to the current temperature, or hopefully a more-sophisticated one which will parse the XML file to set the dialplan variable. The end goal is to have Asterisk play the speech files temperature sixty five degrees or the equivalent non-English files per the channel's current language setting. Thank you. Any assistance will be greatly appreciated. Sincerely, Trevor Hammonds ___ -- 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] AGI to announce temperature from weather.com XML file
Check this one out. developed for AstLinux, it ought to be close to what you want. depending on your version, you may need to modify sound file references http://lonnie.abelbeck.com/astlinux/info/weather.php John Novack Trevor Hammonds wrote: I would like to have the ability to have Asterisk announce the temperature -- not using TTS -- within the dialplan. For a non-Asterisk project, I have a cron job that periodically pulls down an XML file from weather.com containing local weather data (TWC's user agreement requires that data be cached locally). Using sed, I also create a text file that contains only the numeric value of the current temperature, created from that XML file (e.g. tmp65/tmp in the XML file becomes a text file with 65 as its only contents). I am hoping someone on the list has an example of a lightweight AGI script that I may modify to either read the simple text file and set a dialplan variable to the current temperature, or hopefully a more-sophisticated one which will parse the XML file to set the dialplan variable. The end goal is to have Asterisk play the speech files temperature sixty five degrees or the equivalent non-English files per the channel's current language setting. Thank you. Any assistance will be greatly appreciated. Sincerely, Trevor Hammonds ___ -- 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 -- Dog is my co-pilot ___ -- 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] Sending things to Jabber but not within anextension
Each of these should be do-able either through dialplan snippets, cron jobs or AGI's. (a) would be a dialplan snippet (b) would be a dialplan snippet (c) would require an AGI or cron to monitor how long the peer has been out of service. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Reynolds Sent: Thursday, July 16, 2009 5:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sending things to Jabber but not within anextension I have set my Asterisk server up to connect to my Jabber server and send messages with the caller ID details in them to the recipients of incoming calls - this is working very nicely. There are a few other things I can think of right now that I would like to send to Jabber but as yet I do not know whether they are possible. They are: (a) a count of messages in a voicemail box - triggered when the user connects to Jabber. (b) a notification that a voicemail was left - triggered when the caller leaving the message hangs up. (c) a notification that a specific SIP peer has become unreachable - or better still, has been unreachable for five minutes. As I say, I do not know if any of these things are possible, nor how to do them if they are. I have looked at what is on voip-info.org and what the book has but have not seen anything that seems relevant. Of course, I may be right that these are not possible at present, or I may have missed or misunderstood how to do it. I am on version 1.4.21.2~dfsg-1-pmr-2 - which is from Debian stable plus a locally added patch. Thanks in advance to anyone who can advise, even if just to confirm that it is not possible yet. -- Phil Reynolds mail: phil-aster...@tinsleyviaduct.com Web: http://www.tinsleyviaduct.com/phil/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 This message was sent using IMP, the Internet Messaging Program. ___ -- 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] AGI to announce temperature from weather.com XML file
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Trevor Hammonds wrote: I am hoping someone on the list has an example of a lightweight AGI script that I may modify to either read the simple text file and set a dialplan variable to the current temperature, or hopefully a more-sophisticated one which will parse the XML file to set the dialplan variable. I think that the 'FILE' function will do what you're looking for. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKXyZsCFu3bIiwtTARAg/rAJ9D6RQQ2N51GNU8sWHbxPyJM82U1ACgn4bR c55n6BEUTuMPRSsRgETeE9w= =YLth -END PGP SIGNATURE- ___ -- 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] Sending faxes with T.38 problem. Fax for Asterisk (no SpanDSP) - 1.6.1.1
I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax
(.tiff) from the first
asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates an
INVITE with audio G.711. Asterisk2
accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with T.38 to
Asterisk1. But, Asterisk1 responds with
488 not acceptable here. I double check t38pt_udptl = yes in my sip.conf. Why
not Asterisk1 can not accept the Re-INVITE
with T.38 SDP? What do I miss?
dev10*CLI fax show version
Fax For Asterisk Components:
dev10*CLApplications: 1.6.1_1.0.11
dev10*CLDigium Fax T.38 Driver: 1.6.1_1.0.9 (optimized for i686_32)
dev10*CLDigium Fax G.711 Driver: 1.6.1_1.0.9 (optimized for i686_32)
--
.call file
Channel: SIP/1...@outbound-calls
MaxRetries: 3
WaitTime: 30
Set: LOCALSTATIONID=2
Set: LOCALHEADERINFO=T38 fax
Set: T38CALL=1
Set: T38TXDETECT=yes
CallerID: 123456
Context: fax-tx
Extension: send
priority:1
---
sip.conf
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
t38pt_udptl = yes
extensions.conf
[fax-rx]
exten = receive,1,NoOp( FAX RECEIVE )
exten = receive,n,Set(GLOBAL(FAXCOUNT)=${GLOBAL(FAXCOUNT)}+1)
exten = receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten = receive,n,Set(FAXFILE=fax-${FAXCOUNT}-rx.tif)
exten = receive,n,Set(GLOBAL(LASTFAXCALLERNUM)=${CALLERID(num)})
exten = receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)})
exten = receive,n,NoOp( SETTING FAXOPT )
exten = receive,n,Set(FAXOPT(ecm)=yes)
exten = receive,n,Set(FAXOPT(headerinfo)=MY FAXBACK RX)
exten = receive,n,Set(FAXOPT(localstationid)=1234567890)
exten = receive,n,Set(FAXOPT(maxrate)=14400)
exten = receive,n,Set(FAXOPT(minrate)=2400)
exten = receive,n,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten = receive,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten = receive,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten = receive,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten = receive,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten = receive,n,NoOp( RECEIVING FAX : ${FAXFILE} )
exten = receive,n,ReceiveFAX(/home/sip/fax/${FAXFILE})
[fax-tx]
exten = send,1,NoOp( SENDING FAX )
exten = send,n,Wait(6)
exten = send,n,Set(GLOBAL(FAXCOUNT)=1)
;exten = send,n,Set(GLOBAL(FAXCOUNT)= ${GLOBAL(FAXCOUNT)}+1)
exten = send.,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten = send,n,Set(FAXFILE=test.tif)
; Set FAXOPTs
exten = send,n,NoOp( SETTING FAXOPT )
exten = send,n,Set(FAXOPT(ecm)=yes)
exten = send,n,Set(FAXOPT(headerinfo)=Fax from ${GLOBAL(LASTFAXCALLERNAME)} at
${GLOBAL(LASTFAXCALLERNUM)} was received.)
exten = send,n,Set(FAXOPT(localstationid)=1234567890)
exten = send,n,Set(FAXOPT(maxrate)=14400)
exten = send,n,Set(FAXOPT(minrate)=2400)
; Send the fax
exten = send,n,NoOp( SENDING FAX : ${FAXFILE} )
exten = send,n,SendFAX(/home/sip/fax/${FAXFILE},d)
[default]
exten = _X.,1,NoOp( FAX DETECTED )
exten = _X.,n,Goto(fax-rx,receive,1)
--
The SIP trace is
#
U 2009/07/15 22:30:11.588436 74.13.233.143:5060 - 209.167.0.151:5060
INVITE sip:1...@209.167.0.151 SIP/2.0..Via: SIP/2.0/UDP
74.13.233.143:5060;branch=z9hG4bK092e48ce;rport..Max-Forwards: 70..From
: 123456 sip:123...@74.13.233.143;tag=as74992a24..To:
sip:1...@209.167.0.151..Contact: sip:123...@74.13.233.143..Call-I
D: 422fd4375fe79a5977e891870f5cc...@74.13.233.143..cseq: 102
INVITE..User-Agent: Asterisk PBX 1.6.1.1..Date: Wed, 15 Jul 2009 2
2:30:11 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY..Supported: replaces, timer..Content-Type: appl
ication/sdp..Content-Length: 265v=0..o=root 1425900082 1425900082 IN IP4
74.13.233.143..s=Asterisk PBX 1.6.1.1..c=IN IP4 74
.13.233.143..t=0 0..m=audio 18452 RTP/AVP 0 101..a=rtpmap:0
PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=si
lenceSupp:off - - - -..a=ptime:20..a=sendrecv..
#
U 2009/07/15 22:30:11.723006 209.167.0.151:5060 - 74.13.233.143:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
74.13.233.143:5060;branch=z9hG4bK092e48ce;received=74.13.233.143;rport=5060..From:
123456
sip:123...@74.13.233.143;tag=as74992a24..To:
sip:1...@209.167.0.151..Call-ID: 422fd4375fe79a5977e891870f5cc...@74.13.233.
143..CSeq: 102 INVITE..Server: Asterisk PBX 1.6.1.1..Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Suppor
ted: replaces, timer..Contact: sip:1...@209.167.0.151..Content-Length:
0
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10) - Solved.
On Wed, Jul 15, 2009 at 5:23 PM, Waynewa...@planetwayne.com wrote: Hi all, Just a quickie to say that this has been solved now - real simple - downloaded '*current*' rather than the versions from the home page of Astrisk.org. (didn't realise there was a 'current' version tbh. Anyways - I don't get Asterisk seg faulting now when hammering the speaker button on my cisco phones :) Interestingly - I've got another query - but will post another question when I've had chance to play more. Cheers Wayne. Wayne wrote: Hi all, I've just built a new installation of CentOS release 5.3 (Final) and have installed both 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 Huh? http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz is not the same as http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.10.tar.gz ? Their sha1 files are identical. sean ___ -- 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] DEVICE_STATE() and Asterisk 1.6.0.10
Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mark Michelson wrote:
You need to set a call-limit for the SIP peer. Device state calculation for
a
SIP peer is predicated on both the call-limit and busylevel. Let's say that
you
were to have a call-limit of 2, but no busylevel set. These are the device
states reported for the peer based on the number of calls currently handled:
Hi Mark. Thanks for your explanation of these parameters.
I should have posted my configurations. I double-checked the contents
of sip.conf and I have this. The 'subscribecontext' was added for
testing, per the other reply I got for my question.
;
; Settings common to all devices on our system
;
[basic-options](!)
type=friend
host=dynamic
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
qualify=yes
;
; Standard desksets here
;
[lan-deskset](!,basic-options)
context=sip-deskset
notifyringing = yes
notifyhold = yes
limitonpeers = yes
call-limit=99
[6668](lan-deskset)
secret=mysecret
callerid=Matts SIP 6668
username=Barry's IP450
call-limit=32
busylevel=1
subscribecontext=hint-context
My hint-context is:
[hint-context]
exten = 6668,hint,SIP/6668;
I'm still not getting anything other than NOT_INUSE from DEVICE_STATE.
Here is the CLI output:
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:1]
NoOp(SIP/-0955ecc8, SIP/6668 has state NOT_INUSE) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:2]
NoOp(SIP/-0955ecc8, SIP/ has state NOT_INUSE) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:3]
ExecIf(SIP/-0955ecc8, 0?Busy(10)) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:4]
Dial(SIP/-0955ecc8, SIP/6668) in new stack
And here is sip show inuse:
corp-asterisk*CLI sip show inuse
* User name In use Limit
6668 1 32
6667 0 99
1 99
* Peer name In use Limit
6668 1/1/0 32
6667 0/0/0 99
0/0/0 99
For completeness, here is the dialplan that's producing this:
exten = 6668,1,NoOp(SIP/${EXTEN} has state ${DEVICE_STATE(SIP/${EXTEN})});
exten = 6668,n,NoOp(SIP/ has state ${DEVICE_STATE(SIP/)});
exten = 6668,n,ExecIf($[${DEVICE_STATE(SIP/${EXTEN})}=BUSY]?Busy(10));
exten = 6668,n,Dial(SIP/${EXTEN});
Thanks for the config info. I have a couple of suggestions for fixes.
1. Try changing the type in [basic-options] from friend to peer. I've found
that
device state reporting for outbound calls (from the perspective of the phone)
tends to be more accurate with this type.
2. If for some odd reason number 1 either doesn't sound appealing to you or
doesn't work, then try moving the limitonpeers=yes option from your
[basic-options] section to the [general] section.
No, neither of these ideas actually make any real sense to me, but they are
based on behavior that I have witnessed with my Asterisk setup in my office.
Mark Michelson
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Re: [asterisk-users] Sending faxes with T.38 problem. Fax for Asterisk (no SpanDSP) - 1.6.1.1
hutx wrote: I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax (.tiff) from the first asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates an INVITE with audio G.711. Asterisk2 accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with T.38 to Asterisk1. But, Asterisk1 responds with 488 not acceptable here. I double check t38pt_udptl = yes in my sip.conf. Why not Asterisk1 can not accept the Re-INVITE with T.38 SDP? What do I miss? You need to post a console log from the Asterisk console of this problem occurring, with 'core set verbose 10', 'core set debug 10' and 'sip set debug on' (and ensure that the 'debug' logger level is activated for the console log channel in logger.conf). -- 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] H323 situation
Hi all, I have this installation: Asterisk 1.6.1.1 with h323 support, pwlib_v1_10_3 and openh323_v1_18_0. I have a problem that is, when a call comes from H323 and goes to a Sip phone the asterisk sends two rtp streams to the sip. I checked this with tcpdump, save the payload (voice is in G711u), one is the ringing indication and the other is the voice coming from the user in h323 side. And worst they go to the same port. This causes that in the sip phone there are problems, when the call is answered sometimes we get the riging indication, others a mix of the two with very bad sound quality and others(few) a god audio call. The outgoing calls from sip to H323 are ok. I also tested an incoming call from a dahdi channel and from here everything is ok, only one rtp stream and a good call. By the way I had other problem that I fixed, but don't know if it was in the best way. The h323 box is a Cisco AS5300 (or 5350?) and when I was making outgoing calls the AS disconnected all of them after 10 sec. I investigated I noticed that the AS as a limitation to the G711 payload to 20 ms, and asterisk was using 150 ms. I resolve this changing in frame.c the codec value and recompile asterisk. There is simpler way to do this? Like changing values in codec.conf?... Regards LS ___ -- 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] Voicemail login incorrect
Hi all, I'm having trouble with voicemail on my *NOW 1.5/FreePBX box. I have enabled voicemail in the extensions area, and set the default password. However, every time I try to log in with a mailbox and password, I get the message login incorrect. I've tried changing the voicemail password, and also disabling and re-enabling the voicemail feature. What else can I do to set up the voicemail? Also, I've left the VM Context as default and the mailbox is 101. Thanks! --Zaheer ___ -- 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] Suggestions for web based soft phones
Hi Zeeshan, You might want to take a look at our solution here: http://www.flashsip.com/ We do the customization of the software for our clients on demande. Best regards, Brian On Sat, Jul 11, 2009 at 2:55 AM, Zeeshan Zakaria zisha...@gmail.com wrote: For a while now I've been looking for a good web based soft phone solution, but so far no luck. A few solutions which I've tried, both Java based and Flash based, either don't work, or had bad sound quality. I need something which I could put on my productions server for my clients. Seems like good web based solutions are all paid ones, nobody is giving it for free. Any ideas, suggestions whom to go with? Thanks -- Zeeshan A Zakaria ___ -- 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] DEVICE_STATE() and Asterisk 1.6.0.10
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mark Michelson wrote: Thanks for the config info. I have a couple of suggestions for fixes. 1. Try changing the type in [basic-options] from friend to peer. I've found that device state reporting for outbound calls (from the perspective of the phone) tends to be more accurate with this type. 2. If for some odd reason number 1 either doesn't sound appealing to you or doesn't work, then try moving the limitonpeers=yes option from your [basic-options] section to the [general] section. No, neither of these ideas actually make any real sense to me, but they are based on behavior that I have witnessed with my Asterisk setup in my office. Mark Michelson I'll give these a try and see if they help. At this point, I'd be willing to slaughter a goat and place its entrails on the keyboard if I thought it would help. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKX0w5CFu3bIiwtTARAsy8AKCDbPMDZJ98v1HuL/KLDuQsayI84ACfX4OI Jw5YOgQllm1+wbq2wThh4Wg= =eE0m -END PGP SIGNATURE- ___ -- 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] T38 negotiation, the last step !
Hi, I've managed to get HYLAFAXT38MODEM-ASTERISKCISCOAS5400 working, but when they are negotiating asterisk drops a message telling Unknown RTP codec 96 received from gateway Do somebody know how to fix it ? Thank you ! [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [Jul 16 17:50:39] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [Jul 16 17:50:40] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [Jul 16 17:50:41] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] ___ -- 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] Voicemail login incorrect
On Thu, 2009-07-16 at 10:57 -0400, Zaheer Master wrote: Hi all, I'm having trouble with voicemail on my *NOW 1.5/FreePBX box. I have enabled voicemail in the extensions area, and set the default password. However, every time I try to log in with a mailbox and password, I get the message login incorrect. I've tried changing the voicemail password, and also disabling and re-enabling the voicemail feature. What else can I do to set up the voicemail? Also, I've left the VM Context as default and the mailbox is 101. snip If you set your Asterisk console to a verbose mode, what password do you see passed to the voicemail application? We recently noticed 3CX softphone users with multiple options set for DTMF were sending duplicate DTMF signals to our voicemail resulting in the same problems you are seeing, e.g., 1234 would be sent as 11223344. Just a guess - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Voicemail login incorrect - SOLVED
Thanks for the reply John. In the voicemail.conf file there were two extra [] creating a NULL context. Removing those extra brackets fixed the problem. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, July 16, 2009 12:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: 'AsteriskNOW Discussion' Subject: Re: [asterisk-users] Voicemail login incorrect On Thu, 2009-07-16 at 10:57 -0400, Zaheer Master wrote: Hi all, I'm having trouble with voicemail on my *NOW 1.5/FreePBX box. I have enabled voicemail in the extensions area, and set the default password. However, every time I try to log in with a mailbox and password, I get the message login incorrect. I've tried changing the voicemail password, and also disabling and re-enabling the voicemail feature. What else can I do to set up the voicemail? Also, I've left the VM Context as default and the mailbox is 101. snip If you set your Asterisk console to a verbose mode, what password do you see passed to the voicemail application? We recently noticed 3CX softphone users with multiple options set for DTMF were sending duplicate DTMF signals to our voicemail resulting in the same problems you are seeing, e.g., 1234 would be sent as 11223344. Just a guess - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] T38 negotiation, the last step !
Xavier Cardil wrote: Hi, I've managed to get HYLAFAXT38MODEM- ASTERISKCISCOAS5400 working, but when they are negotiating asterisk drops a message telling Unknown RTP codec 96 received from gateway Do somebody know how to fix it ? There's nothing to fix; the gateway sent an expected RTP packet, which Asterisk dropped. If your FAX works, then you can ignore this. -- 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
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10) - Solved.
Huh? http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz is not the same as http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.10.tar.gz? Their sha1 files are identical. sean I believe he means that: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz is not the same as svn checkout http://svn.digium.com/svn/asterisk/branches/1.6.0 Which is true as there are lots of things that have been fixed in the Subversion repo. -Jonathan ___ -- 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] Mexican ITSP needed
Try http://www.inext.com.mx they can provide DIDs in several cities in Mexico. On Thu, 2009-07-16 at 09:16 +0200, Michiel van Baak wrote: Hey all, I was wondering if anyone knows about a Mexican ITSP I can connect to to route calls from and to my * boxen. If it matters: I'm located in The Netherlands and one of our customers is in Mexico so if we need a Mexican presence that is not an issue. Thanks. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 negotiation, the last step !
Xavier Cardil escribió: Hi, I've managed to get HYLAFAXT38MODEM- ASTERISKCISCOAS5400 working, but when they are negotiating asterisk drops a message telling Unknown RTP codec 96 received from gateway Do somebody know how to fix it ? Thank you ! [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [Jul 16 17:50:39] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [Jul 16 17:50:40] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [Jul 16 17:50:41] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] It looks like the Cisco gateway is trying to negotiate or send some video codec or something which asterisk does not recognize. According to the spec (http://www.iana.org/assignments/rtp-parameters), The codecs 96 to 127 are dynamic RTP payloads that are negotiated with another protocols to define them. If everything else on your fax session works fine, this message shouldn't be a problem. I think this is much like the X-lite 126 codec type message: [Jul 16 11:35:57] NOTICE[10989]: rtp.c:1287 ast_rtp_read: Unknown RTP codec 126 received from '0.0.0.0' And doesn't pose a problem anyway. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- 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] T38 negotiation, the last step !
Hi Kelvin, thank you for your response, well in fact it is not working but that's only a NOTICE, not an error. Warnings comes after that and the fax is not sent. Take a look at the last lines of this output : [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-01ba5638] IVR3*CLI debug channel allSIP/GWCISCO5400O-01ba5638 SIP/T38modem-01ba2178 [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/GWCISCO5400O-01ba5638] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ] [SIP/GWCISCO5400O-01ba5638] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] [Jul 16 18:20:20] NOTICE[2846]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] [Jul 16 18:20:21] NOTICE[2846]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] [Jul 16 18:20:21] NOTICE[2846]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] [Jul 16 18:20:30] WARNING[2815]: chan_sip.c:3075 retrans_pkt: Maximum retries exceeded on transmission 1af1c92e-9270-de11-8264-001517bb9...@ivr3for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. [Jul 16 18:20:30] WARNING[2815]: chan_sip.c:3102 retrans_pkt: Hanging up call 1af1c92e-9270-de11-8264-001517bb9...@ivr3 - no reply to our critical packet (see doc/sip-retransmit.txt). Do you have an idea of what is happening ? I sniffed UDP traffic on port 5060 and I get : INVITE SDP ( t38 ) -- TRYING -- NOT ACCEPTABLE HERE ACK BYE--- 200 OK Thanks for your help. On Thu, Jul 16, 2009 at 6:17 PM, Kevin P. Fleming kpflem...@digium.comwrote: Xavier Cardil wrote: Hi, I've managed to get HYLAFAXT38MODEM- ASTERISKCISCOAS5400 working, but when they are negotiating asterisk drops a message telling Unknown RTP codec 96 received from gateway Do somebody know how to fix it ? There's nothing to fix; the gateway sent an expected RTP packet, which Asterisk dropped. If your FAX works, then you can ignore this. -- 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 graph1 Description: Binary data ___ -- 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] Polycom Spectralink 8002 WiFi Phones
Michael Graves wrote: On Tue, 14 Jul 2009 22:58:14 + (UTC), Jeff LaCoursiere wrote: Jeff, yeah i saw the posts, i followed Bob Pierce config and had no luck, BUT it just started to work, i changed AP's, seems like theres something wrong with Ubiquiti NanoStation2 WMM implementation, i used a Linksys WRT54G2 and viola! it started to work, i guess i should've done that to begin with... :( I'll play around whit the Nanostations QoS settings and see if i can get it to work on those AP's. What AP's were you using? Hi Cesar, I did actually get it to work as well, and was using Linksys WRT54G with dd-wrt. I *intended* for the phone to be useful at random wifi hotspots, however, and was a bit disappointed to find that that was not going to work. So it sits on a shelf gathering dust... I had one of these for evalution last spring. The resulting review is here: http://www.smallnetbuilder.com/content/view/30498/80/ They work well enough when paired with suitable APs. According to Polycom you must support WMM or all bets are off. In my case I had a Netgear WRT-2000. I had no issues at all with integration with Asterisk. However, I think that there truly isn't a dedicated Wifi handset that will satisfy if you want to be able to roam the world and make calls from public hotspots. Too many hotspots require a web login before you get access. Thus the best devices for this sort of thing seem to be Nokia dual mode phones with built-in web browsers. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- 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 Michael, actually the intended use for this phones is at a customers warehouse and its trailer/truck yard, not roaming around public hotspots, so its now going to come down to selecting the proper AP's as you suggest, we were looking at Naonstation2 for their WDS implemetation, so maybe a set of linksys AP's with dd-wrt will do the job. I've been testing it with this Linksys WRT54G2 since yesterday and works great with asterisk. Great review btw. -Cesar ___ -- 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] Iphone setup
Thank you for the heads up. I will look into both weephone and voipover3g ___ -- 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] early-dial SIP 484 incomplete address, dialplan patterns and international calls
Hi, I would like to know if someone can suggest me an efficient way of writing a dialplan to match variable-length international calls when using SIP clients with the early dial or 484 feature. What I usually do for clients that do NOT early dial is define something like this in my outbound context: For local calls (they fortunately have a fixed length): exten = _0Z, ... For international numbers (variable length: exten = _000ZX., ... etc. However, as you can expect, early dial phones will work fine for local calls but will fail for international numbers because Asterisk will start dialing out as soon as it matches, say, 00044, and ignore the digits sent afterwards. What would be the most efficient dialplan solution for a mixed client environment. Should my _000ZX. logic have to wait for digits until timeout? Suggestions greatly appreciated. Vieri ___ -- 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] iax.conf, IP-based access control
The documentation in http://svn.digium.com/svn/asterisk/branches/1.4/configs/iax.conf.sample (and http://svn.digium.com/svn/asterisk/branches/1.6.*/configs/iax.conf.sample) seems slightly wrong. --- ; ... Limited IP based ; access control is allowed by use of allow and deny keywords. ... --- allow specifies an allowed codec. It should read: --- ; ... Limited IP based ; access control is allowed by use of permit and deny keywords. ... --- Codecs: disallow/allow Netmasks: deny/permit I think this does not justify filing a bug. 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] iax.conf, IP-based access control
On Thursday 16 July 2009 12:26:25 Philipp Kempgen wrote: I think this does not justify filing a bug. No, it does. Go ahead and file it. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- 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] QoS
At 06:37 AM 7/15/2009, you wrote: Ours is just internal, but the concept should be the same. My boss could talk on his phone fine until he cranked up Foxnews feed. Once the video started, he couldn't talk on his phone anymore (bad quality or total loss of call). What I've done here is probably a bit extreme, but we've never had a problem of any kind with our VOIP calls. it's a house, no more than 2 calls at a time on cable internet so it might just be that the connection is significantly faster than we ever use. I have a Linksys router that has the Asterisk box connected to a port marked High and the rest of the house is a second router connected to a port on the first router flagged as low or regular. I ran separate Cat5 for the phones and the computers. If I knew what I was doing I'd get a Linux box with 3 ports and have it do everything. Ira ___ -- 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] AGI to announce temperature from weather.com XML file
On Thu, 2009-07-16 at 04:49 -0700, Trevor Hammonds wrote: I would like to have the ability to have Asterisk announce the temperature -- not using TTS -- within the dialplan. Chapter 9 of Asterisk: The Future of Telephony shows you how to build an AGI script to do just that. For a free download, check out www.asteriskdocs.org. There are obviously many other ways to do it. -- Jared Smith Training Manager 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] iax.conf, IP-based access control
Tilghman Lesher schrieb: On Thursday 16 July 2009 12:26:25 Philipp Kempgen wrote: I think this does not justify filing a bug. No, it does. Go ahead and file it. ok. https://issues.asterisk.org/view.php?id=15518 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] PRI hunt group
I managed to do it with a simple forward to the one of our DIDs The problem now is that I loose the CID of the original caller ... Is there a way to forward the call with the original caller ID ? Thanks! G. On Wed, Jul 15, 2009 at 5:38 PM, John Novack jnov...@stromberg-carlson.orgwrote: Steve Totaro wrote: Forwarding a POTS line will not work, it is like a trunk to trunk transfer so it is not free, so the line stays busy. You need to port that number over to the PRI provider. That all depends on the POTS provider. Multiple calls from one POTS number CAN be done, but finding anyone these days in a business office that can do anything the least bit out of the ordinary is near impossible, and one may not be willing to pay for it. Same with repair - The stock answer - We checked the line from here, it has to be your equipment even with an open pair in their ( unmaintained ) Outside Plant is about all one can expect. Also, not all POTS lines are portable. In the US some ILEC's are exempt from the provisions of the 1996 telecom Act. All above assume this is in the US. John Novack On Wed, Jul 15, 2009 at 7:41 PM, Gondar Monn gonda...@gmail.com mailto:gonda...@gmail.com wrote: Thank you for your quick answers! @ Brent: rollover is on, I would like to any calls that come on 5551234 to another DID, to be able to receive several calls on the same number @ Don: You are right, I am talking about a specific DID: We have an analog line with busy forward setup @ the telco to forward calls to 5551234 ... But 2 lines are not enough, so I would like to do the same locally: All calls to 555-1234 are forwarded to 555-2345 to free up the line Does that make sense ? To answer your second question: calls to 555-1234 are alerted on the first channel available channel. All subsequent calls to the DID report busy Again, Thanks for helping me out G. On Wed, Jul 15, 2009 at 4:21 PM, Don Kelly d...@donkelly.biz mailto:d...@donkelly.biz wrote: Rollover or hunting is generally the default on PRIs. It sounds like Gondar's concern is with a specific DID number (Do multiple calls to other DID numbers work OK?). I'd wonder about a couple things: Are people dialing '5551234' directly, or are calls being forwarded to that number? Some call-forwarding schemes will only forward one call at a time, giving other callers a busy signal. Are you sure that calls to '5551234' aren't being alerted on more than one channel and your Asterisk configuration is presenting the busy? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson Sent: Wednesday, July 15, 2009 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI hunt group Gondar Monn wrote: I am having trouble with a DID on a PRI. If there is a call to that DID (let say 5551234) , the next calls get a busy signal. How to I go about sending the call to the next available channel ? Thanks! G. If the telco is providing the PRI then you need to tell them you want rollover on the PRI's. Otherwise, anybody calling across the PSTN to the DID number that is bound to the PRI channel is going to get a busy signal from the telco if that channel is in use. ___ -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 -- Dog is my co-pilot ___ -- 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
Re: [asterisk-users] setvar and transfer
Philipp Kempgen philipp.kemp...@amooma.de writes: Benny Amorsen schrieb: Last concern: Does setvar work even for transfers, like accountcode does? I can't answer your question, but transfer != transfer. Some use a feature code in Asterisk, some initiate a transfer on their phone, some use a way to call the Transfer() application. Mixing it up causes a lot of confusion. Sorry, I meant Does setvar work even for SIP blind transfers during calls, like accountcode codes? /Benny ___ -- 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] PRI hunt group
Looks like the caller ID gets lost when you forward. Normally if you do a simple forward using central office features, the caller ID will be the calling party's number. If you're using a PBX (or something that looks like one) the PBX does a hook-flash, makes a call to your PRI DID and you see the number of the line doing the forwarding. As someone pointed out earlier, port it if you can. --Don _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gondar Monn Sent: Thursday, July 16, 2009 1:56 PM To: novacks...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI hunt group I managed to do it with a simple forward to the one of our DIDs The problem now is that I loose the CID of the original caller ... Is there a way to forward the call with the original caller ID ? Thanks! G. On Wed, Jul 15, 2009 at 5:38 PM, John Novack jnov...@stromberg-carlson.org wrote: Steve Totaro wrote: Forwarding a POTS line will not work, it is like a trunk to trunk transfer so it is not free, so the line stays busy. You need to port that number over to the PRI provider. That all depends on the POTS provider. Multiple calls from one POTS number CAN be done, but finding anyone these days in a business office that can do anything the least bit out of the ordinary is near impossible, and one may not be willing to pay for it. Same with repair - The stock answer - We checked the line from here, it has to be your equipment even with an open pair in their ( unmaintained ) Outside Plant is about all one can expect. Also, not all POTS lines are portable. In the US some ILEC's are exempt from the provisions of the 1996 telecom Act. All above assume this is in the US. John Novack On Wed, Jul 15, 2009 at 7:41 PM, Gondar Monn gonda...@gmail.com mailto:gonda...@gmail.com wrote: Thank you for your quick answers! @ Brent: rollover is on, I would like to any calls that come on 5551234 to another DID, to be able to receive several calls on the same number @ Don: You are right, I am talking about a specific DID: We have an analog line with busy forward setup @ the telco to forward calls to 5551234 ... But 2 lines are not enough, so I would like to do the same locally: All calls to 555-1234 are forwarded to 555-2345 to free up the line Does that make sense ? To answer your second question: calls to 555-1234 are alerted on the first channel available channel. All subsequent calls to the DID report busy Again, Thanks for helping me out G. On Wed, Jul 15, 2009 at 4:21 PM, Don Kelly d...@donkelly.biz mailto:d...@donkelly.biz wrote: Rollover or hunting is generally the default on PRIs. It sounds like Gondar's concern is with a specific DID number (Do multiple calls to other DID numbers work OK?). I'd wonder about a couple things: Are people dialing '5551234' directly, or are calls being forwarded to that number? Some call-forwarding schemes will only forward one call at a time, giving other callers a busy signal. Are you sure that calls to '5551234' aren't being alerted on more than one channel and your Asterisk configuration is presenting the busy? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson Sent: Wednesday, July 15, 2009 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI hunt group Gondar Monn wrote: I am having trouble with a DID on a PRI. If there is a call to that DID (let say 5551234) , the next calls get a busy signal. How to I go about sending the call to the next available channel ? Thanks! G. If the telco is providing the PRI then you need to tell them you want rollover on the PRI's. Otherwise, anybody calling across the PSTN to the DID number that is bound to the PRI channel is going to get a busy signal from the telco if that channel is in use. ___ -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation
[asterisk-users] possible to configure 2 servers - one is backup system for the other?
Hello everybody! Please let me ask you a question: Is it possible (and if yes, how) to configure 2 asterisk servers on two machines so that the second one acts as a backup system if the first one is unresponsive? Clearly, the second should take over automacigally (but not necessarily during an active call). Many thanks for your time! 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] possible to configure 2 servers - one is backup system for the other?
- Norbert Zawodsky norb...@zawodsky.at wrote: Hello everybody! Please let me ask you a question: Is it possible (and if yes, how) to configure 2 asterisk servers on two machines so that the second one acts as a backup system if the first one is unresponsive? Clearly, the second should take over automacigally (but not necessarily during an active call). Many thanks for your time! Norbert Check out Linux-HA using heartbeat and a disk replication system like DRBD. Solid. --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] setvar and transfer
I'm going to give a qualified no. The reason being is that setvar works in a session (say SIP/100-abcdefg) and the blind transfer may spawn a new session like Local/1-abcdefg). So your only solid variables are the global ones. You can verify this by looking at CLI output with verbose set to at least 5. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen Sent: Thursday, July 16, 2009 2:18 PM To: Philipp Kempgen Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] setvar and transfer Philipp Kempgen philipp.kemp...@amooma.de writes: Benny Amorsen schrieb: Last concern: Does setvar work even for transfers, like accountcode does? I can't answer your question, but transfer != transfer. Some use a feature code in Asterisk, some initiate a transfer on their phone, some use a way to call the Transfer() application. Mixing it up causes a lot of confusion. Sorry, I meant Does setvar work even for SIP blind transfers during calls, like accountcode codes? /Benny ___ -- 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] How to Change size of CDR(accountcode) variable?
On Tue, 2009-07-14 at 00:01 +0200, Benny Amorsen wrote: Last concern: Does setvar work even for transfers, like accountcode does? At least in theory, the setvar= setting in sip.conf or iax.conf (or in Asterisk 1.6.0 and later, chan_dahdi.conf) should work just like the Set() dialplan application, in that you can prepend an underscore or two to the variable name to make it inheritable by spawned channels. So, in theory, setvar=_FANCYLONGACCOUNTCODE=foo should make that channel variable inheritable by the *next* spawned channel (but not any channels beyond that), and setvar=__FANCYLONGACCOUNTCODE=foo should make it inheritable by the spawned channel *and* any channels it spawns, and so forth. That being said, it's just theory. I have not tested this in my lab, but I offer it as a simple suggestion for you to try. Please let us know if this helped. (My gut feeling is that it should work for DTMF and flash-based transfers. I'm a little less sure about SIP-initiated transfers.) -- Jared Smith Training Manager 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
[asterisk-users] 800 number portability
Apologies for the off topic post... hoping someone knows if 800 number portability in the states is legally enforced? One of my customers is being told by their current vanity 800 provider that they own the number and refuse to release it to their new carrier. I thought I understood that in 1991 the FCC mandated portability by 1993. Are they bluffing? They want a 3 year buyout to release the number! j ___ -- 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] Unique id used for call recording missing from CDR data for transferred call
Hello,
I have an application that needs to record outgoing calls. It's
running on Asterisk 1.4.18, with CDR data stored in MySQL.
Outgoing calls are recorded based on their uniqueid. When outgoing
calls are placed, there is a line like this on my extensions.conf:
exten = _.,n,MixMonitor(/var/spool/asterisk/monitor/${UNIQUEID}.gsm)
For regular outgoing calls, this works fine. The call is recorded in
a file named for its uniqueid, and if I need the recording I can pull
the information out of the CDR table, find the uniqueid, then pull up
the recording. We have a Web application that does this
automatically.
When an incoming is transfered to an outgoing call via an attended SIP
transfer, however, the uniqueid assigned to the outgoing call does not
seem to end up in the CDR table at all.
Here's what happens:
1. User A calls in to the queue on Zap/1-1 uniqueid 1247686911.203
2. Operator picks up call.
3. Queue application starts recording in 1247686911.203.gsm
4. Operator talks for awhile, decides to transfer call
5. Operator switches Line 2, calls User B on channel Zap/24-1 with
uniqueid 1247686911.205
6. Dialplan command MixMonitor(1247686911.205.gsm) starts recording
7. Operator talks to User B for awhile
8. Operator transfers call, connects User A to User B, and hangs up
This generates these 4 CDRs:
1. 2009-07-15 15:41:51
channel Zap/1-1
dstchannel SIP/DEV-0078ebd0
duration 175
uniqueid 1247686911.203
2. 2009-07-15 15:43:17
channel SIP/DEV-0078ebd0
dstchannel (none)
duration 89
uniqueid 1247686997.204
3. 2009-07-15 15:43:47
channel Zap/24-1
dstchannel Zap/1-1
duration 80
uniqueid 1247687027.206
4. 2009-07-15 15:43:47
channel SIP/DEV-c4018700ZOMBIE
dstchannel Zap/24-1
duration 59
uniqueid 1247686911.203
The uniqueid for the outgoing call, 1247686911.205, doesn't end up in
any of the CDRs, so is effectively lost.
Any ideas on how to handle this? Is this something that's likely to
be fixed in a later version of Asterisk?
Thanks,
Scott.
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[asterisk-users] Stop recording on SIP attended transfer
Hello, We have an application where operators will sometimes take an incoming call from a queue, then contact an outside line, do a consultation, and finally do a SIP attended transfer to join the two parties together. We'd like to record the incoming caller's conversation with the operator and the attended part of the outgoing call, but not the unattended part, after the transfer has completed and the incoming caller is talking on the outgoing line (this part of the call may be confidential). We start the recording with a MixMonitor command when the outgoing call is placed. However, I don't see anything in the dialplan that gets run when the SIP attended transfer happens, where I could issue a command to stop recording. Any suggestions? Thanks! Scott. ___ -- 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] Stop recording on SIP attended transfer
I don't know the full details, but I think if the Dial command(s) have the W and/or w options on them, you can activate/deactivate recording via DTMF. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Scott Gifford Sent: Thursday, July 16, 2009 4:45 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Stop recording on SIP attended transfer Hello, We have an application where operators will sometimes take an incoming call from a queue, then contact an outside line, do a consultation, and finally do a SIP attended transfer to join the two parties together. We'd like to record the incoming caller's conversation with the operator and the attended part of the outgoing call, but not the unattended part, after the transfer has completed and the incoming caller is talking on the outgoing line (this part of the call may be confidential). We start the recording with a MixMonitor command when the outgoing call is placed. However, I don't see anything in the dialplan that gets run when the SIP attended transfer happens, where I could issue a command to stop recording. Any suggestions? Thanks! Scott. ___ -- 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] 800 number portability
On Thu, Jul 16, 2009 at 5:39 PM, Jeff LaCoursiere j...@jeff.net wrote: Apologies for the off topic post... hoping someone knows if 800 number portability in the states is legally enforced? One of my customers is being told by their current vanity 800 provider that they own the number and refuse to release it to their new carrier. I thought I understood that in 1991 the FCC mandated portability by 1993. Are they bluffing? They want a 3 year buyout to release the number! j It may depend on the way the vanity number was obtained. I bought mine from tollfreenumbers.com or whatever and they are mine with docs to back it up. Perhaps if obtained in another fashion, the fine print may say differently. More explanation of where the numbers came from and any contractual obligations may help. Similarly, there used to be and probably still are places that would allow you to register a domain name for free or close to it. In the fine print was the fact that the domain name was not yours, they registered it on your behalf, in their name. A year later you got a renewal bill or some such and if you wanted to move it, you had to pay through the nose. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] 800 number portability
Changing toll-free RespOrgs (Responsible Organizations) is different from number portability. That said, the owner of a toll-free number has the right to change RespOrgs, so the question is Who is the owner? Has your customer been buying simple toll-free service and owned the number all along, or are they buying some sort of enhanced service and the provider owns the number? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Thursday, July 16, 2009 4:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 800 number portability Apologies for the off topic post... hoping someone knows if 800 number portability in the states is legally enforced? One of my customers is being told by their current vanity 800 provider that they own the number and refuse to release it to their new carrier. I thought I understood that in 1991 the FCC mandated portability by 1993. Are they bluffing? They want a 3 year buyout to release the number! j ___ -- 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] PRI hunt group
Try to get a level one tech to set RDNIS on your forwarded POTS line. Good luck! On Thu, Jul 16, 2009 at 3:30 PM, Don Kelly d...@donkelly.biz wrote: Looks like the caller ID gets lost when you forward. Normally if you do a “simple forward” using central office features, the caller ID will be the calling party’s number. If you’re using a PBX (or something that looks like one) the PBX does a hook-flash, makes a call to your PRI DID and you see the number of the line doing the forwarding. As someone pointed out earlier, port it if you can. --Don -- R *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gondar Monn *Sent:* Thursday, July 16, 2009 1:56 PM *To:* novacks...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] PRI hunt group I managed to do it with a simple forward to the one of our DIDs The problem now is that I loose the CID of the original caller ... Is there a way to forward the call with the original caller ID ? Thanks! G. On Wed, Jul 15, 2009 at 5:38 PM, John Novack jnov...@stromberg-carlson.org wrote: Steve Totaro wrote: Forwarding a POTS line will not work, it is like a trunk to trunk transfer so it is not free, so the line stays busy. You need to port that number over to the PRI provider. That all depends on the POTS provider. Multiple calls from one POTS number CAN be done, but finding anyone these days in a business office that can do anything the least bit out of the ordinary is near impossible, and one may not be willing to pay for it. Same with repair - The stock answer - We checked the line from here, it has to be your equipment even with an open pair in their ( unmaintained ) Outside Plant is about all one can expect. Also, not all POTS lines are portable. In the US some ILEC's are exempt from the provisions of the 1996 telecom Act. All above assume this is in the US. John Novack On Wed, Jul 15, 2009 at 7:41 PM, Gondar Monn gonda...@gmail.com mailto:gonda...@gmail.com wrote: Thank you for your quick answers! @ Brent: rollover is on, I would like to any calls that come on 5551234 to another DID, to be able to receive several calls on the same number @ Don: You are right, I am talking about a specific DID: We have an analog line with busy forward setup @ the telco to forward calls to 5551234 ... But 2 lines are not enough, so I would like to do the same locally: All calls to 555-1234 are forwarded to 555-2345 to free up the line Does that make sense ? To answer your second question: calls to 555-1234 are alerted on the first channel available channel. All subsequent calls to the DID report busy Again, Thanks for helping me out G. On Wed, Jul 15, 2009 at 4:21 PM, Don Kelly d...@donkelly.biz mailto:d...@donkelly.biz wrote: Rollover or hunting is generally the default on PRIs. It sounds like Gondar's concern is with a specific DID number (Do multiple calls to other DID numbers work OK?). I'd wonder about a couple things: Are people dialing '5551234' directly, or are calls being forwarded to that number? Some call-forwarding schemes will only forward one call at a time, giving other callers a busy signal. Are you sure that calls to '5551234' aren't being alerted on more than one channel and your Asterisk configuration is presenting the busy? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson Sent: Wednesday, July 15, 2009 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI hunt group Gondar Monn wrote: I am having trouble with a DID on a PRI. If there is a call to that DID (let say 5551234) , the next calls get a busy signal. How to I go about sending the call to the next available channel ? Thanks! G. If the telco is providing the PRI then you need to tell them you want rollover on the PRI's. Otherwise, anybody calling across the PSTN to the DID number that is bound to the PRI channel is going to get a busy signal from the telco if that channel is in use.
[asterisk-users] Compilation error
Hi all I'm trying to install asteris 1.4.22.1 on Solaris 10...the server is V120 SUN spark...During compilation (gmake) I got the following error /vis.c -o np/vis.o_a np/vis.c: In function `svis': np/vis.c:205: error: `u_int32_t' undeclared (first use in this function) np/vis.c:205: error: (Each undeclared identifier is reported only once np/vis.c:205: error: for each function it appears in.) np/vis.c:205: error: syntax error before u_char np/vis.c:205: error: syntax error before ')' token np/vis.c:205: error: syntax error before u_char np/vis.c:205: error: syntax error before ')' token np/vis.c:207: error: syntax error before u_char np/vis.c:207: error: syntax error before ')' token np/vis.c:207: error: syntax error before u_char np/vis.c:207: error: syntax error before ')' token np/vis.c: In function `strsvis': np/vis.c:245: error: `u_int32_t' undeclared (first use in this function) np/vis.c:245: error: syntax error before u_char np/vis.c:245: error: syntax error before ')' token np/vis.c:245: error: syntax error before u_char np/vis.c:245: error: syntax error before ')' token np/vis.c:248: error: syntax error before u_char np/vis.c:248: error: syntax error before ')' token np/vis.c:248: error: syntax error before u_char np/vis.c:248: error: syntax error before ')' token np/vis.c: In function `strsvisx': np/vis.c:275: error: `u_int32_t' undeclared (first use in this function) np/vis.c:275: error: syntax error before u_char np/vis.c:275: error: syntax error before ')' token np/vis.c:275: error: syntax error before u_char np/vis.c:275: error: syntax error before ')' token np/vis.c:280: error: syntax error before u_char np/vis.c:280: error: syntax error before ')' token np/vis.c:280: error: syntax error before u_char np/vis.c:280: error: syntax error before ')' token np/vis.c: In function `vis': np/vis.c:303: error: `u_int32_t' undeclared (first use in this function) np/vis.c:303: error: syntax error before u_char np/vis.c:303: error: syntax error before ')' token np/vis.c:303: error: syntax error before u_char np/vis.c:303: error: syntax error before ')' token np/vis.c:305: error: syntax error before u_char np/vis.c:305: error: syntax error before ')' token np/vis.c:305: error: syntax error before u_char np/vis.c:305: error: syntax error before ')' token gmake[2]: *** [np/vis.o_a] Error 1 gmake[1]: *** [editline/libedit.a] Error 2 gmake: *** [main] Error 2 I guess there is a BUG in vis.c...Can you please let me know how to define this variable? Thanks ___ -- 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] 800 number portability
On Thu, 16 Jul 2009, Don Kelly wrote: Changing toll-free RespOrgs (Responsible Organizations) is different from number portability. That said, the owner of a toll-free number has the right to change RespOrgs, so the question is Who is the owner? The owner in this case is CallSource (www.callsource.com). Funny enough, it looks a lot like the kind of stuff you do, Don ;) So I guess my disconnect is that a party can own an 800 number, but have it routed by the RespOrg of their choice? In this case my client must be renting the 800 number from CallSource, and they are the actual owner, so are refusing to let it go. Does that sound right? Has your customer been buying simple toll-free service and owned the number all along, or are they buying some sort of enhanced service and the provider owns the number? I assumed it was simple 800 service (and in fact at first they told me it was ATT they were getting the service from). It seems that this is actually something enhanced. Cheers, j ___ -- 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] 800 number portability
There are national number rental agencies that lease out prime 800 numbers even down to the rate center level. They own the number, not the renter, and there is a contract that says so. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Thursday, July 16, 2009 7:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 800 number portability On Thu, 16 Jul 2009, Don Kelly wrote: Changing toll-free RespOrgs (Responsible Organizations) is different from number portability. That said, the owner of a toll-free number has the right to change RespOrgs, so the question is Who is the owner? The owner in this case is CallSource (www.callsource.com). Funny enough, it looks a lot like the kind of stuff you do, Don ;) So I guess my disconnect is that a party can own an 800 number, but have it routed by the RespOrg of their choice? In this case my client must be renting the 800 number from CallSource, and they are the actual owner, so are refusing to let it go. Does that sound right? Has your customer been buying simple toll-free service and owned the number all along, or are they buying some sort of enhanced service and the provider owns the number? I assumed it was simple 800 service (and in fact at first they told me it was ATT they were getting the service from). It seems that this is actually something enhanced. Cheers, j ___ -- 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] Stop recording on SIP attended transfer
Danny Nicholas da...@debsinc.com writes: I don't know the full details, but I think if the Dial command(s) have the W and/or w options on them, you can activate/deactivate recording via DTMF. Thanks, that's a good idea, I might be able to rig something up with that! Scott. ___ -- 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] 2 Problems with 1.6.2
I've been using 1.6.2 for a few weeks and I've managed to get almost everything working perfectly. I can't get the MWI indicators on my Aastra phones to work properly, the did in all the versions of 1.2 I used up to the most recent one, but now they work correctly right after the phone is re-started and rarely thereafter. it's as if something changed in the way the MWI is handled and I can't figure out what the difference is or what I've done wrong. When I restart the machines I can't make an outgoing DAHDI call until I get an incoming call on that same line. Ira ___ -- 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] PRI hunt group
In the good old days telcos didn't care how many channels your forward used up, they just did it. However nowadays they only allow one channel at a time to be forwarded, if you need more you have to pay for it. Verizon here in NJ charges around $8.00 a month for each call path (channel), and so do most CLECs. Optimum cable used to not restrict how many channels, however the last customer I tried this with only one channel worked, even though for the ones I set it up before they started blocking it it still works. If you don't want to port it to the PRI for whatever reason you can convert it to a RCFW (remote call forwarded number) which is around $15.00 plus $8.00 for each additional channel again pricing is for here in Verizon land. On Wed, Jul 15, 2009 at 7:41 PM, Gondar Monngonda...@gmail.com wrote: Thank you for your quick answers! @ Brent: rollover is on, I would like to any calls that come on 5551234 to another DID, to be able to receive several calls on the same number @ Don: You are right, I am talking about a specific DID: We have an analog line with busy forward setup @ the telco to forward calls to 5551234 ... But 2 lines are not enough, so I would like to do the same locally: All calls to 555-1234 are forwarded to 555-2345 to free up the line Does that make sense ? To answer your second question: calls to 555-1234 are alerted on the first channel available channel. All subsequent calls to the DID report busy Again, Thanks for helping me out G. On Wed, Jul 15, 2009 at 4:21 PM, Don Kelly d...@donkelly.biz wrote: Rollover or hunting is generally the default on PRIs. It sounds like Gondar's concern is with a specific DID number (Do multiple calls to other DID numbers work OK?). I'd wonder about a couple things: Are people dialing '5551234' directly, or are calls being forwarded to that number? Some call-forwarding schemes will only forward one call at a time, giving other callers a busy signal. Are you sure that calls to '5551234' aren't being alerted on more than one channel and your Asterisk configuration is presenting the busy? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson Sent: Wednesday, July 15, 2009 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI hunt group Gondar Monn wrote: I am having trouble with a DID on a PRI. If there is a call to that DID (let say 5551234) , the next calls get a busy signal. How to I go about sending the call to the next available channel ? Thanks! G. If the telco is providing the PRI then you need to tell them you want rollover on the PRI's. Otherwise, anybody calling across the PSTN to the DID number that is bound to the PRI channel is going to get a busy signal from the telco if that channel is in use. ___ -- 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 ___ -- 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] 2 Problems with 1.6.2
On Thursday 16 July 2009 20:19:07 Ira wrote: I've been using 1.6.2 for a few weeks and I've managed to get almost everything working perfectly. I can't get the MWI indicators on my Aastra phones to work properly, the did in all the versions of 1.2 I used up to the most recent one, but now they work correctly right after the phone is re-started and rarely thereafter. it's as if something changed in the way the MWI is handled and I can't figure out what the difference is or what I've done wrong. When I restart the machines I can't make an outgoing DAHDI call until I get an incoming call on that same line. It would probably be best for you to read UPGRADE-1.4.txt, UPGRADE-1.6.txt, and UPGRADE.txt, as the issue with MWI is explained in there and what you can do to fix it. The second file contains the explanation, although you would be well advised to read all three. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- 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] 800 number portability
On Thu, 16 Jul 2009, Don Kelly wrote: Changing toll-free RespOrgs (Responsible Organizations) is different from number portability. That said, the owner of a toll-free number has the right to change RespOrgs, so the question is Who is the owner? The owner in this case is CallSource (www.callsource.com). Funny enough, it looks a lot like the kind of stuff you do, Don ;) So I guess my disconnect is that a party can own an 800 number, but have it routed by the RespOrg of their choice? In this case my client must be renting the 800 number from CallSource, and they are the actual owner, so are refusing to let it go. Does that sound right? Has your customer been buying simple toll-free service and owned the number all along, or are they buying some sort of enhanced service and the provider owns the number? I assumed it was simple 800 service (and in fact at first they told me it was ATT they were getting the service from). It seems that this is actually something enhanced. Cheers, j Yes, someone can own a number and transfer service from one RespOrg to another to save a few bucks or get better or different service. If your client originally got the number from CallSource, it probably belongs to CallSource. If they got it from someone else and transferred to CallSource they would have a different argument. At www.callsource.com/services/numbers.php CallSource says Own your tracking numbers. CallSource is one of the few call tracking vendors that will transfer permanent ownership of numbers to you. I think the reasonable person would rely on this statement and not expect they would want three-years' payment for it. Does your customer have their original contract? --Don ___ -- 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] Mexican ITSP needed
On 11:39, Thu 16 Jul 09, Carlos Chavez wrote: Try http://www.inext.com.mx they can provide DIDs in several cities in Mexico. Thanks. I asked the customer to have a look (I'm only capable of reading English and Dutch ;)) You have any experience with them ? On Thu, 2009-07-16 at 09:16 +0200, Michiel van Baak wrote: Hey all, I was wondering if anyone knows about a Mexican ITSP I can connect to to route calls from and to my * boxen. If it matters: I'm located in The Netherlands and one of our customers is in Mexico so if we need a Mexican presence that is not an issue. Thanks. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] PRI hunt group
C F wrote: If you don't want to port it to the PRI for whatever reason you can convert it to a RCFW (remote call forwarded number) which is around $15.00 plus $8.00 for each additional channel again pricing is for here in Verizon land. Is that true even if the number is out of a rate center that is billed long-distance relative to the destination (but still intra-LATA)? Or do you pay normal LD rates on top of all that in the intra-LATA LD scenario? -- Alex Balashov 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
