Re: [asterisk-users] Multiple TDM400p cards in one machine -- no longer an issue?
Hi Chris, we tried TDM800 and TDM2400 without problems even if it is a pity not to have leds on the card (as TDM400). BTW I think it is better to have only one card on PBXs, when possible of course!! Giorgio Chris Earle wrote: Hi all, Years ago, I was pretty sure attempting to use two TDM400p cards in one machine was recommended against by Digium ... probably because the cards couldn't hack it, and/or interrupt problems etc I have seen some posts recently that seem to indicate it is in fact possible these days thanks to some updated firmware perhaps? . I just need to have two in the server because the 4 ports aren't enough ... I'd rather just expand by one card rather than get a TDM2400 (or TDM800??) Anyone had recent success/failure with this sort of thing? -- Chris Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] proper permissions for asterisk and it's spooler?
Can someone tell me real quick what the file and directory permissions are for asterisk and especially the spooler? Executing [EMAIL PROTECTED]:1] DeadAGI(SIP/UXMC-0914dcc8, postqueue.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/postqueue.agi + SOX=sox + SOXMIX=soxmix + SRCFRMT=wav + DESTFRMT=wav ++ ls '/var/spool/asterisk/monitor/*out.*' ls: /var/spool/asterisk/monitor/*out.*: No such file or directory Please! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SBC
I am trying to make a mirroring for my asterisk using nextone SBC,I have a problem ,which is when and end point send Invitation to SBC realm . This realm is send INV and REG messages to Asterisk. Asterisk sends INV message again to this realm. NexTone SBC try to send again to asterisk and this is caused loop. There solution was , Asterisk should send to a different realm of NexTone or different GW. How can I do that from asterisk.(define a signaling ip) Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!
after 18 hours, over 200 pages of reading, a complete reinstall of asterisk
I am down to this.
extensions.conf
[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1
[default]
exten = 8005181896,1,Dial,(IAX2/UXMC)
exten = s,1,Answer()
(I tried)
exten = _1XX,1,DIAL,(IAX2/teliax/${EXTEN},30,tr)
(as well)
iax.conf
[general]
port=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
register = :[EMAIL PROTECTED]
[teliax]
context=default
type=friend
host=voip-co3.teliax.com
auth=md5
user=
secret=x
disallow=all
allow=ulaw
allow=alaw
allow=gsm
sip.conf
[UXMC]
user=xxx
context=internal
type=friend
qualify=yes
nat=no
secret=
canreinvite=no
host=dynamic
nat=no
If I put back previous config, I can call into the 1800 number and here that
silly chick heckle me from my server!
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[asterisk-users] Allow for context includes in realtime (ARA)
Hi list, still wondering for a couple of days how to handle context includes in realtime architecture i've tried to patch my pbx_realtime.c with a patch on the digium issue tracker (http://bugs.digium.com/view.php?id=6014) but it does not seem to work or may be i'm using the wrong way does anyone has a solution thanks for reply BR -- Cheikhou DIAW ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring Asterisk as Gateway SIP-H.323 via ooh323
Hi, I'm trying to configure Asterisk as SIP-H.323 Gateway via ooh323, but I have an error relatively to the GK Confirmation message. From the log: H323 RAS channel creation - succesful Sent GRQ message Gatekeeper Confirmed (GCF) message received ERROR:No Gatekeeper ID present in received GKconfirmed message Ignoring message and will retransmit GRQ after timeout Error: Failed to handle received RAS message This is my ooh323.conf file: [general] port=1720 bindaddr=130.177.137.214 ;the address of my Asterisk server h323id=ObjSysAsterisk e164=100100 callerid=asterisk gateway=yes gatekeeper = 192.57.108.112 ;the address of my Tandberg GK faststart=yes h245tunneling=yes context=default disallow=all allow=gsm allow=ulaw allow=alaw dtmfmode=rfc2833 [myuser1] type=user context=context1 disallow=all allow=gsm allow=ulaw [mypeer1] type=peer context=context2 ip=a.b.c.d ; UPDATE with appropriate ip address port=1720 ; UPDATE with appropriate port e164=101 [myfriend1] type=friend context=default ip=a.b.c.d ; UPDATE with appropriate ip address port=1720 ; UPDATE with appropriate port disallow=all allow=ulaw e164=12345 rtptimeout=60 dtmfmode=rfc2833 Anyone can help me? Regards Dino -- Dino Anaclerio email: [EMAIL PROTECTED] art: http://dnacl.deviantart.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple host= in sip.conf
Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw canreinvite=no [vz3] context=outbound type=friend host=.z.z.z.z disallow=all allow=alaw canreinvite=no However, is there anyway I can have just one account for him, with mult host= statements, so I can authenticate him based on his IP in just one place? -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fax2mail ann missing CallerID number
Hello.
I have a problem recieving fax without a callerid. Somehow the script i'm
using fails and i don't know how to fix it. Does anyone have an idea how to
solve this? Here an example of a working fax transmission:
fax2mail v2.0
Triggered on Tuesday, May 29 2007, at 10:38 AM
$1 = CallerID number of fax sender = 02365207150
$2 = CallerID name of fax sender =
$3 = Fax number called = FaxNum
$4 = Destination name = RecipName
$5 = Destination email address = [EMAIL PROTECTED]
$6 = Fax file name (without .tif extension)
= /var/spool/asterisk/fax/02365207150
$7 = Format conversion (n=none,p=pdf,e=eps) = p
Fax file /var/spool/asterisk/fax/02365207150.tif found.
Converted /var/spool/asterisk/fax/02365207150.tif
to /var/spool/asterisk/fax/02365207150.pdf.
E-mailed file to [EMAIL PROTECTED]
Removing destination file /var/spool/asterisk/fax/02365207150.pdf
And here without the transmitted callerid:
fax2mail v2.0
Triggered on Wednesday, May 30 2007, at 10:04 AM
$1 = CallerID number of fax sender =
$2 = CallerID name of fax sender = FaxNum
$3 = Fax number called = RecipName
$4 = Destination name = [EMAIL PROTECTED]
$5 = Destination email address = /var/spool/asterisk/fax/
$6 = Fax file name (without .tif extension) = p
$7 = Format conversion (n=none,p=pdf,e=eps) =
Fax file p.tif not found.
E-mailed warning to /var/spool/asterisk/fax/
In this case i have a file called .tif in my /var/spool/asterisk/fax folder.
Here the relevant part of asterisk extension.conf
exten = 49,1,Set(FAXFILE=/var/spool/asterisk/fax/${CALLERIDNUM}.tif)
exten = 49,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERIDNUM})
exten = 49,n,rxfax(${FAXFILE})
exten = 49,n,System('/usr/local/bin/fax2mail ${CALLERIDNUM} ${CALLERIDNAME}
FaxNum RecipName [EMAIL PROTECTED] ${FAXFILENOEXT} p')
Thanks in advance for any help.
best regards
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Re: [asterisk-users] multiple host= in sip.conf
I dont think asterisk supports this . You can have host=dynamic and he can send calls from different servers . Problem will arise when you need to call him ( if registrations are enabled then latest registration will be getting call from you or you can directly send calls to his ip . ) On 30/05/07, Yusuf [EMAIL PROTECTED] wrote: Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw canreinvite=no [vz3] context=outbound type=friend host=.z.z.z.z disallow=all allow=alaw canreinvite=no However, is there anyway I can have just one account for him, with mult host= statements, so I can authenticate him based on his IP in just one place? -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!
Can you post some output from asterisk cli output while you make call ?
On 30/05/07, BSumrall [EMAIL PROTECTED] wrote:
after 18 hours, over 200 pages of reading, a complete reinstall of asterisk
I am down to this.
extensions.conf
[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1
[default]
exten = 8005181896,1,Dial,(IAX2/UXMC)
exten = s,1,Answer()
(I tried)
exten = _1XX,1,DIAL,(IAX2/teliax/${EXTEN},30,tr)
(as well)
iax.conf
[general]
port=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
register = :[EMAIL PROTECTED]
[teliax]
context=default
type=friend
host=voip-co3.teliax.com
auth=md5
user=
secret=x
disallow=all
allow=ulaw
allow=alaw
allow=gsm
sip.conf
[UXMC]
user=xxx
context=internal
type=friend
qualify=yes
nat=no
secret=
canreinvite=no
host=dynamic
nat=no
If I put back previous config, I can call into the 1800 number and here
that silly chick heckle me from my server!
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Re: [asterisk-users] multiple host= in sip.conf
Not supported jsut use host=dynamic with username and secret. Alex Yusuf a écrit : Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw canreinvite=no [vz3] context=outbound type=friend host=.z.z.z.z disallow=all allow=alaw canreinvite=no However, is there anyway I can have just one account for him, with mult host= statements, so I can authenticate him based on his IP in just one place? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Dial Command to a non-Asterisk url
This is what is shown when the call connects with: sip show channel The conference suite from another provider on internal IP is waiting for an ACK on port 5605, but * is sending it back to port 2289 Internal between Asterisk and another Conference suite: * SIP Call Direction: Outgoing Call-ID:[EMAIL PROTECTED] Our Codec Capability: 14 Non-Codec Capability: 1 Their Codec Capability: 4 Joint Codec Capability: 4 Format ulaw Theoretical Address:192.168.45.183:5605 Received Address: 192.168.45.183:2289 NAT Support:Always Audio IP: 192.168.45.196 (local) Our Tag:as31c610d6 Their Tag: t1122b SIP User agent: Username: slee Peername: slee Original uri: sip:[EMAIL PROTECTED]:5605 Need Destroy: 0 Last Message: Tx: ACK Promiscuous Redir: No Route: sip:[EMAIL PROTECTED]:5605 DTMF Mode: rfc2833 SIP Options:(none) Inbound from SIP Provider: * SIP Call Direction: Incoming Call-ID:[EMAIL PROTECTED] -- REMOVED Our Codec Capability: 14 Non-Codec Capability: 1 Their Codec Capability: 14 Joint Codec Capability: 14 Format gsm Theoretical Address:193.111.201.32:5060 Received Address: 193.111.201.32:5060 NAT Support:Always Audio IP: xx.xx.xx.xx (local) -- REMOVED Our Tag:as65c31c43 Their Tag: as26378dd7 SIP User agent: Asterisk PBX Original uri: sip:[EMAIL PROTECTED] -- REMOVED Caller-ID: 01X -- REMOVED Need Destroy: 0 Last Message: Rx: ACK Promiscuous Redir: No Route: sip:193.111.201.32;lr=on;ftag=as26378dd7 DTMF Mode: rfc2833 SIP Options:(none) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] False ring problem
Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R and r options in Dial application but they dont work. -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple host= in sip.conf
Thing is, he does not REGISTER to me, he just uses me as proxy for his calls. I authenticate his calls in his IP. Alexandre VERNIOL wrote: Not supported jsut use host=dynamic with username and secret. Alex Yusuf a écrit : Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw canreinvite=no [vz3] context=outbound type=friend host=.z.z.z.z disallow=all allow=alaw canreinvite=no However, is there anyway I can have just one account for him, with mult host= statements, so I can authenticate him based on his IP in just one place? -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Tuesday, May 29, 2007 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW * John covici wrote: I have an install using Rhino cards -- I sure hope they get their act together by then. They have no choice now, do they? Nothing focuses the attention like a deadline. -Stephen- You are lucky to have any support for anything other than Adtran. The very early Asterisk code supported Adtran only. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk as a call recorder for ISDN30 ?
It would be possible but you should check out Oreka. It works great if all you need is recording. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mike Dent Sent: Tuesday, May 29, 2007 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk as a call recorder for ISDN30 ? Hi, would it be possible to use Asterisk to record calls only? There would be an existing PBX and calls come in on a ISDN30 line? The Asterisk box would need to sit between the incoming ISDN 30 circuit and the existing PBX. Is this possible? thanks Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW *
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Tuesday, May 29, 2007 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW * I think its a fair decision . 1.2 is very stable and they are not closing it all together , security issues will still be fixed . They need to concentrate more on 1.4 to make it bugfree . Fair indeed. I would guess that a completely stable 1.2 w/ security maintenance is acceptable to the majority of users. Those folks still using 1.0.x certainly aren't clamoring for new features! The great many folks using 1.2 are happy w/ a stable release and don't necessarily need new features. A lot of those folks might consider moving to 1.4 when the stability issues and bugs are worked out. Possibly there are features that they would like to have but they don't want to invest the time and effort into a migration until they are reasonably confident that 1.4 will meet their needs. I think that having the development team be able to focus the majority of their attention on improving 1.4 is better than having them split their time between the old and new releases. I'm feeling like there's more ROI to be had improving 1.4. -MC I do hope that when they find major security bugs like the recent SIP bug for example, that affected both 1.2.x and 1.4.x, they backport the fix. At least if the code base has not changed all that much and it is only a few lines of code. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please advise: Asterisk on Dell PowerEdge 1750 w/ hyperthreading
Hi Gordon, Any reason you mention Debian? Is it just preference? Or...? I'm a Debian person, myself. We have * 1.2 on a PowerEdge 2950 with CentOS (which I dislike). We got it commercially done and it's under warranty, although I don't think they'd mind if we muck around with the OS - they'd probably just charge if they have to do some servicing. I'd make sure I have system image backups. If I had my way / spare time / etc I'd put Debian... but for the risk of wrecking a perfectly working, production system. bu Gordon Henderson wrote: On Tue, 29 May 2007, Zeeshan Zakaria wrote: Anyone else with any suggestions? Hard to work out what to suggest - what's your expected load going to be? Any telco cards? etc. If you want support from Dell, then it's RHEL whatever... Personally, if I had that hardware, I'd load up Debian Etch, compile up a custom kernel for it, compile up the latest asterisk (rather than use the Debian package), and just get on with it. Hyperthreading shouldn't be an issue with a relatively recent kernel (it doesn't appear to be for me, but my load is relatively light on the HT processor I'm using - 2-300 calls a day at most right now) and if you don't like HT, you can always turn it off. Gordon On 5/28/07, Kapil Dhawan [EMAIL PROTECTED] wrote: Redhat Enterprise Zeeshan Zakaria wrote: I've to setup Asterisk on a Dell PowerEdge 1750 server. Its dual Xeon 3GHz with Hyperthreading. People on this list who have experience with this server please advise me how is the performance of Asterisk on this server, what flavour of linux is good on it etc. Is Hyperthreading going to be a problem or not. I once read somewhere that hyperthreading caused some voice quality problems in Asterisk. Is it fixed in or not yet? Any other suggestions will also be helpful. Thanks -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] False ring problem
Do you have 'r' in ur dial command ? 'r' makes asterisk produce ring . On 30/05/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R and r options in Dial application but they dont work. -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax2mail ann missing CallerID number
Thomas Stein wrote: Hello. I have a problem recieving fax without a callerid. Somehow the script i'm using fails and i don't know how to fix it. Does anyone have an idea how to It looks like the script expects a caller-id number and uses that for the .tif name. You'll need to update your script and check to see if $1 is blank. If it is then assign it a value. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] False ring problem
Rizwan Hisham wrote: Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk Show use that section of your dial plan. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
We still run 1.2.6 on some of our production systems because, so far, it has been the only stable release of Asterisk for us. Other versions core dump for no reason and do all kinds of other funky things. On 5/29/07, Jaswinder Singh [EMAIL PROTECTED] wrote: What you say might be true for small business or home pbx systems . But if you have a production server handling sip/iax trunks over internet then you need to upgrade to avoid security related bugs and exploits that are released . You seem to miss the idea here. You work with a version that supports your feature needs and find the sub-version that provides the most stability for your deployments. Lets face it these boxes should go in and run for weeks, months or even years without much intervention (assuming the mission of the box does not change). I'm running a 1.2.7.something (i think) that has been running almost nonstop since installing. Very reliable and stable for my needs. Compared to a Merlin or Nortel or any other system out that I feel I have a much better product. Could I benefit from a newer sub-version? Maybe. Will I upgrade the box in it current roll? No. Unless the application I use the box for has a major change (or the hardware dies) I'll just let it keep on running as it is. For my future deploys I am working closer with 1.4. The reason is clear. 1.4 is the future of asterisk. When 1.6 or 2.0 comes out I'll investigate into migrating in that direction at that time because that will become the future of asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple host= in sip.conf
for inbound connections how does asterisk manage host=host-name returning multiple A records... will it allow authentication for any of the IP's returned? I would assume that in the case of 'inbound' if you specify a host- name that you have PTR records for you could do it in one entry again I'm making a blind assumption. IE say you 10.23.23.3, .4, .5 as his IP's if you created entries either in your own dns or etc hosts (depending on os) you should be able to create entries for each of his IP's all resolving to the same name... and then one entry ... for his transactions from him - you. now the reverse of you - him you would in theory loose control over which host you send the call to but if he doesn't care then it wold work... and while this assumes you have no moral / security objection to using host-names. someone would have to keep my honest here though as I haven't looked at where asterisk does the NS lookup and how those transactions work. if it only read the conf file and did a translation at startup via a single lookup for host name then this wouldn't work. On May 30, 2007, at 6:11 AM, Yusuf wrote: Thing is, he does not REGISTER to me, he just uses me as proxy for his calls. I authenticate his calls in his IP. Alexandre VERNIOL wrote: Not supported jsut use host=dynamic with username and secret. Alex Yusuf a écrit : Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw canreinvite=no [vz3] context=outbound type=friend host=.z.z.z.z disallow=all allow=alaw canreinvite=no However, is there anyway I can have just one account for him, with mult host= statements, so I can authenticate him based on his IP in just one place? -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations Phone: 703-944-9909 -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call transfer while dialing
Hi, I want to transfer the call to a conferencing room while dialing. I tried to do that using manager API(Redirect), but it did't work. Regards, Jason. Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] False ring problem
You should (must!) remove any r/R parameter from your command. If you do that, no false ring will be generated anymore... Att, Ricardo. Rizwan Hisham escreveu: Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R and r options in Dial application but they dont work. -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please advise: Asterisk on Dell PowerEdge 1750 w/ hyperthreading
On Wed, 30 May 2007, [EMAIL PROTECTED] wrote: Hi Gordon, Any reason you mention Debian? Is it just preference? Or...? I've been using it since 0.96r1 ... ie. about 12-13 years?!? I have many LAMP type servers running stuff which are as rock solid as a rock type thing... Eg. the server I'm typing this email on: 13:16:13 up 398 days, 3:23, 2 users, load average: 0.06, 0.12, 0.10 another: 13:16:33 up 587 days, 23:18, 1 user, load average: 0.00, 0.00, 0.00 anod another - note the load average )-: 13:22:19 up 90 days, 23:39, 1 user, load average: 3.57, 3.99, 3.17 (that's a very busy vBulletin server - just say no!) But I don't always use Debian packages - I tend to treat it as the base system, then if the supplied packages aren't what I need, then I compile them myself. This habit stems from years of supporting different systems - SunOs/Solaris, *BSD, HP, IRIX, etc. although I have to say in recent years, it's been almost exclusively Debian only... I always compile up a static kernel tuned to the exact hardware (zaptel, etc. modules are the exception these days) and off I go. That's not for everyone though, but it's a habit I've gotten into I'm a Debian person, myself. We have * 1.2 on a PowerEdge 2950 with CentOS (which I dislike). We got it commercially done and it's under warranty, although I don't think they'd mind if we muck around with the OS - they'd probably just charge if they have to do some servicing. I'd make sure I have system image backups. I've run 2950's as file/dns/nis/samba servers in the past. (for clients) Not tried asterisk on them though. Etch ought to boot OK, Sarge won't boot directly (lack of network/disk drivers in the ageing kernel in Sarge) unless you use one of the Dell specific loaders. I prefer Asus servers for my own stuff. If I had my way / spare time / etc I'd put Debian... but for the risk of wrecking a perfectly working, production system. Quite! If it ain't broke ... (although there are some who might argue that RH/FC/CentOs are broken by design ;-) Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
I'm running a 1.2.7.something (i think) that has been running almost nonstop since installing. Very reliable and stable for my needs. This version has some security issues inside. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asternic Flash panel
Anyone get it working on 1.4. Checked out their website no updates for some time now... -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' Wise men talk because they have something to say; fools, because they have to say something. -- Plato smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] False ring problem
There is no R/r option in my dial application.im only using gM option here
is the dialplan:
exten= _1X.,1,NoOp(Dialing Local!!!)
exten= _1X.,2,Dial(Sip/[EMAIL PROTECTED],,gM
(payasyougo^${CDR(accountcode)}^${CDR(userfield)}))
exten= _1X.,3,Hangup
On 5/30/07, Ricardo Martins [EMAIL PROTECTED] wrote:
You should (must!) remove any r/R parameter from your command. If you do
that, no false ring will be generated anymore...
Att, Ricardo.
Rizwan Hisham escreveu:
Hi all,
when a user dials any number, asterisk automatically generates ringing
which caller can hear, and after 2 - 3 rings asterisk detects that the
called user is busy, then caller hears busy tone. for example user hears---
tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing
at the start so that user hears only beep beep beep if the called user is
busy. I have used the R and r options in Dial application but they dont
work.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
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Re: [asterisk-users] False ring problem
Maybe its a bug in asterisk 1.4.2
On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
There is no R/r option in my dial application.im only using gM option
here is the dialplan:
exten= _1X.,1,NoOp(Dialing Local!!!)
exten= _1X.,2,Dial(Sip/[EMAIL
PROTECTED],,gM(payasyougo^${CDR(accountcode)}^${CDR(userfield)}))
exten= _1X.,3,Hangup
On 5/30/07, Ricardo Martins [EMAIL PROTECTED] wrote:
You should (must!) remove any r/R parameter from your command. If you
do that, no false ring will be generated anymore...
Att, Ricardo.
Rizwan Hisham escreveu:
Hi all,
when a user dials any number, asterisk automatically generates ringing
which caller can hear, and after 2 - 3 rings asterisk detects that the
called user is busy, then caller hears busy tone. for example user hears---
tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing
at the start so that user hears only beep beep beep if the called user is
busy. I have used the R and r options in Dial application but they dont
work.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
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--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
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Software Engineer
AXVOICE Inc.
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Re: [asterisk-users] False ring problem
Here is my CLI output:
Called [EMAIL PROTECTED]
-- SIP/CARRIER-OUT-007d0310 is ringing
-- Call on SIP/CARRIER-OUT-007d0310 left from hold
-- SIP/CARRIER-007d0310 is making progress passing it to
SIP/pepsi-00f267e0
i clearly notice that when the first orange cli msg appears then the actual
ringing starts. like this tone -- tone -- totone -- tone, and if the callee
is busy then tone -- tone -- tobeep beep .
does anyone know what this means: -- Call on SIP/CARRIER-OUT-007d0310 left
from hold
On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Maybe its a bug in asterisk 1.4.2
On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
There is no R/r option in my dial application.im only using gM option
here is the dialplan:
exten= _1X.,1,NoOp(Dialing Local!!!)
exten= _1X.,2,Dial(Sip/[EMAIL
PROTECTED],,gM(payasyougo^${CDR(accountcode)}^${CDR(userfield)}))
exten= _1X.,3,Hangup
On 5/30/07, Ricardo Martins [EMAIL PROTECTED] wrote:
You should (must!) remove any r/R parameter from your command. If you
do that, no false ring will be generated anymore...
Att, Ricardo.
Rizwan Hisham escreveu:
Hi all,
when a user dials any number, asterisk automatically generates ringing
which caller can hear, and after 2 - 3 rings asterisk detects that the
called user is busy, then caller hears busy tone. for example user hears---
tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing
at the start so that user hears only beep beep beep if the called user is
busy. I have used the R and r options in Dial application but they dont
work.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
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--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
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Software Engineer
AXVOICE Inc.
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Re: [asterisk-users] channel_find_locked: Avoided deadlock
I too have this problem. I have two queues set up, and one is in use. I didn't realize thats what caused those errors. I am also using sip. Here are my setups if it helps anyone find a bug: Queues.conf [billing] music=default strategy=ringall reportholdtime = no timeout=8 retry=10 wrapuptime=10 maxlen = 0 announce-frequency = 0 announce-holdtime = no member = Agent/3876 member = Agent/5055 member = Agent/8318 member = Agent/8323 member = Agent/8324 Agents.conf ;Billing agent = 3876,,Christina agent = 8318,,Stephanie agent = 8323,,Rob agent = 8324,,Colleen agent = 5055,,Chris Extensions.conf exten = s,1,Answer() exten = s,n,Ringing() exten = s,n,Wait(2) exten = s,n,Queue(billing,t|||30) exten = s,n,Voicemail(u) exten = s,n,Hangup() ram wrote: On 5/30/07, *Jaswinder Singh* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Is it over iax and there are lot of outgoing channels ? If yes then you are not the only person having this .. SIP ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] False ring problem
It seems that the ring issue is on the CARRIER-OUT signaling. It's
sending you a SIP-Ring-Message and your asterisk-box is sending it to
the callee. The second green line ".is ringing" apears jut because
your box received a ring signal from the CARRIER-OUT. Got the point?
I don't know what the "left from hold" means but seems to be related to
the situation when we push the "flash" button on the phone to put "on
hold" and flash again to put "out of hold". But I'm realy not sure
about it.
Rgds, Ricardo Martins
Rizwan Hisham escreveu:
Here is my CLI output:
Called [EMAIL PROTECTED]
-- SIP/CARRIER-OUT-007d0310
is ringing
-- Call on
SIP/CARRIER-OUT-007d0310 left from hold
-- SIP/CARRIER-007d0310
is making progress passing it to SIP/pepsi-00f267e0
i clearly notice that when the first orange cli msg appears then the
actual ringing starts. like this tone -- tone -- totone -- tone, and if
the callee is busy then tone -- tone -- tobeep beep .
does anyone know what this means: -- Call on SIP/CARRIER-OUT-007d0310
left from hold
On 5/30/07, Rizwan Hisham
[EMAIL PROTECTED] wrote:
Maybe
its a bug in asterisk 1.4.2
On 5/30/07, Rizwan Hisham [EMAIL PROTECTED]
wrote:
There
is no R/r option in my dial application.im
only using gM option here is the dialplan:
exten=
_1X.,1,NoOp("Dialing Local!!!")
exten=
_1X.,2,Dial(Sip/[EMAIL PROTECTED],,gM(payasyougo^${CDR(accountcode)}^${CDR(userfield)}))
exten= _1X.,3,Hangup
On 5/30/07,
Ricardo Martins
[EMAIL PROTECTED] wrote:
You should (must!) remove
any r/R parameter from your command. If you
do that, no false ring will be generated anymore...
Att, Ricardo.
Rizwan Hisham escreveu:
Hi all,
when a user dials any number, asterisk automatically generates ringing
which caller can hear, and after 2 - 3 rings asterisk detects that the
called user is busy, then caller hears busy tone. for example user
hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the
false ringing at the start so that user hears only beep beep beep if
the called user is busy. I have used the R and r options in Dial
application but they dont work.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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--
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--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW *
On 5/30/07, Steve Totaro [EMAIL PROTECTED] wrote: I do hope that when they find major security bugs like the recent SIP bug for example, that affected both 1.2.x and 1.4.x, they backport the fix. At least if the code base has not changed all that much and it is only a few lines of code. Yes, that's the whole idea of putting Asterisk 1.2 into Security Maintenance Mode (or whatever the official name is for it). Security issues will still be fixed for 1.2.x releases, but non-security-related bug fixes will only be applied to the 1.4 branch and trunk. I would anticipate that security issues will continue to be fixed until the next branch (1.6?) is released, and enough time has elapsed until 1.4 is put into security mode as well. The idea is that at any given time, you'll have: Trunk -- new features + bug fixes + security fixes Current release branch -- bug fixes + security fixes, but no new features Previous release branch -- security fixes only (after ~6 months from the date that the current release branch is released). Just to clarify, we currently have: Trunk -- new features + bug fixes + security fixes 1.4 -- bug fixes + security fixes, but no new features 1.2 -- security fixes only When 1.6 is released we'll have: Trunk -- new features + bug fixes + security fixes 1.6 -- bug fixes + security fixes, but no new features 1.4 -- bug fixes + security fixes, but no new features and about six months after 1.6 is released, we'll have: Trunk -- new features + bug fixes + security fixes 1.6 -- bug fixes + security fixes, but no new features 1.4 security fixes only The idea is to give everyone a reasonable amount of time to migrate their systems, after a new release branch is released. I think everyone realizes that a new release branch isn't automagically perfect, and it takes a little time to shake out the bugs. If you're still having problems with the 1.4 branch (or with specific versions of the 1.2 branch), I suggest you do the following to help the developers track down the problems: 1) Check to see if there are any other bug reports with the same symptoms as your own. 2) If there aren't any, fill out your own bug report. Please include as much pertinant information as possible. Does the problem only occur during high call volumes? Is it repeatable? Was a core file generated? If so, please provide a backtrace. 3) Please work with the bug marshalls and developers as they request feedback in the bug tracker. Unfortunately, we have a high number of bugs where someone reports a bug, but doesn't give any additional information when requested. 4) Try any suggested patches. I know this is difficult for some people (especially those who are running Asterisk in production systems, and don't have a test environment setup). Unfortunately, the developers can't fix the bugs without your help. -Jared ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Xen domu with tdm400 hardware
Jonathan Creasy wrote: Which sounds like exactly what I described. Asterisk in Dom0... Whether it's Xen dom0 or domU barely matters. You're still working with a patched kernel. You're taking your chances. Good luck! -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alcatel - Asterisk setup
Following zapata.conf works for us, interconnecting Asterisk - BCM. Never tested with Alcatel though. Jorge Mendoza = Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=es context=from-zaptel signalling=pri_cpe switchtype=qsig rxwink=300 loadzone=pe defaultzone=pe channel = 1-15,17-31 ;for E1 callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=6 callprogress=yes faxdetect=incoming Vieri wrote: According to http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI the author had trouble with QSIG. It would be great if you could give me an extract of your zapata.conf in your successful QSIG setup. And any other tip for that matter. --- Jorge Mendoza [EMAIL PROTECTED] wrote: In my experience, many times Qsig is mandatory for interconnection between Asterisk and others PBX using PRI. Jorge Mendoza Vieri wrote: I'm having the same trouble when the Alcatel-Asterisk trunk has prefix meaning set to open routing number. I enabled overlapdial but still get the same behavior. When it's set to routing number Asterisk receives the full dialed number but it's limited to a maximum of 8 digits. Has anyone solved this open routing number issue that passes only the first digit and ignores the rest? --- Sahil Gupta [EMAIL PROTECTED] wrote: Hi, You need to enable overlapdial. Regards, Sahil Gupta Chief Executive Officer VoiceValley Group of Companies Phone: +61-7-30188403 Fax: +61-7-30188499 On Tue, 29 May 2007, Carlos Hernandez wrote: Hi all: We are looking for someone with experience in Alcatel PBX - PRI - Asterisk integration Please get in touch off list.. We're wanting to hire a professional subcontractor, developer or company to get around some issues like these: Asterisk shows PRI to Alcatel is up, but when trying to dial from Alcatel to Asterisk results in a disc tone (Asterisk do send calls properly into Alcatel) If / when we manage to get anything from Alcatel, we get just the first digit of the number the user is intending to call.. Asterisk expects the whole number at once, so it fails.. Most of the time we get nothing at all from Alcatel, we think something is missing, so Alcatel sees the link is down. Please let me know if you have done this type of work before. We are not wanting to involve the Alcatel people, unless really required. Is there any special way to set up zaptel/zapata so Alcatel detects the PRI to be operational? Is there any special way to receive the calls once the PRI is up? Right now asterisk is set with: pri_net Any information or hints will be greatly appreciated Thank you, Carlos NZ You snooze, you lose. Get messages ASAP with AutoCheck in the all-new Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_html.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem on incoming call from Zap channel to SIP phones...
Carlos Chavez wrote: On Tue, 29 May 2007 20:20:13 -0500, Eric \ManxPower\ Wieling wrote I just made another test by dialing to a Zap channel instead of a SIP phone and the call goes through without any problem. It is just when you try to dial to a SIP phone that you get the auto-congestion message. All other phones in the system are working properly, they are all registered and you can send and receive calls from anywhere except that zap channel. I'm suspicious of the Zap channel in the off-hook state. It should on-hook when on-hook and off-hook when in use. Is that channel still in off-hook? You say you made no changes, it just stopped working. Did anything *else* change around the time this problem appeared? Did someone move a device, or did you update a driver? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Multiple TDM400p cards in one machine -- nolonger an issue?
Chris Earle wrote: Well, yeah, I know it's do-able with either the Sangoma card or Digium's own TDM2400 but I don't want to replace the TDM400p I've already got in there Anyone think two TDM400p's won't cause me any trouble? I think I replied to this already, but I'll give it another go: 1. If your cards are relatively new (last 18 months), and 2. your server mainboard supports IO-APIC (advanced programmable interrupt controller), and 3. your Linux kernel is configured to support SMP (whether or not it has dual processors) and IO-APIC you shouldn't have any problems. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] False ring problem
Do you mean to say that -- first the carrier sends the msg to us to ring and
then the end user sends the msg to ring?
On 5/30/07, Ricardo Martins [EMAIL PROTECTED] wrote:
It seems that the ring issue is on the CARRIER-OUT signaling. It's
sending you a SIP-Ring-Message and your asterisk-box is sending it to the
callee. The second green line .is ringing apears jut because your box
received a ring signal from the CARRIER-OUT. Got the point?
I don't know what the left from hold means but seems to be related to
the situation when we push the flash button on the phone to put on hold
and flash again to put out of hold. But I'm realy not sure about it.
Rgds, Ricardo Martins
Rizwan Hisham escreveu:
Here is my CLI output:
Called [EMAIL PROTECTED]
-- SIP/CARRIER-OUT-007d0310 is ringing
-- Call on SIP/CARRIER-OUT-007d0310 left from hold
-- SIP/CARRIER-007d0310 is making progress passing it to
SIP/pepsi-00f267e0
i clearly notice that when the first orange cli msg appears then the
actual ringing starts. like this tone -- tone -- totone -- tone, and if the
callee is busy then tone -- tone -- tobeep beep .
does anyone know what this means: -- Call on SIP/CARRIER-OUT-007d0310 left
from hold
On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Maybe its a bug in asterisk 1.4.2
On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
There is no R/r option in my dial application.im only using gM option
here is the dialplan:
exten= _1X.,1,NoOp(Dialing Local!!!)
exten= _1X.,2,Dial(Sip/[EMAIL
PROTECTED],,gM(payasyougo^${CDR(accountcode)}^${CDR(userfield)}))
exten= _1X.,3,Hangup
On 5/30/07, Ricardo Martins [EMAIL PROTECTED] wrote:
You should (must!) remove any r/R parameter from your command. If
you do that, no false ring will be generated anymore...
Att, Ricardo.
Rizwan Hisham escreveu:
Hi all,
when a user dials any number, asterisk automatically generates
ringing which caller can hear, and after 2 - 3 rings asterisk detects that
the called user is busy, then caller hears busy tone. for example user
hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the
false
ringing at the start so that user hears only beep beep beep if the called
user is busy. I have used the R and r options in Dial application but they
dont work.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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Re: [asterisk-users] Blindside Web Conferencing
Thanks Stefan and Steve for the links. Will take a look at it. Stefan, thanks for Asterisk-java. Hope to contribute back soon. Glad to know you also got a web conferencing app project. We are using Asterisk-Java, AppFuse (www.appfuse.org), and Icefaces (www.icefaces.org). We're looking at OpenWengo as a desktop client. However, not much work has been done on that yet. Will look into Spark, Openfire, and Asterisk-IM. Back on topic. We have uploaded a WAR file that you can deploy to your appserver (e.g. Tomcat) and test drive. Instructions are at http://www.blindsideproject.org/cgi-bin/trac.cgi/wiki/BlindsideQuickStart Please let us know if you run into problems and feel free to edit the wiki to improve the docs. Thanks. Richard On 5/28/07, Stefan Reuter [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hey Brad, I am not sure if you know about the Asterisk-IM plugin for Openfire. Basically it supports dialing contacts and arbitrary numbers through Spark and updates presence based on being on call or not. One of our next steps would be to integrate conferencing so you could setup (and control) a voice conference much the same way you can do with Jabber groupchat. We also have a web conferencing app in a pre beta state sitting around for some time now (based on Asterisk-Java, DWR and Tomcat) with the original intend to use it for a commercial service which never got really started though. I am not sure if we could come together in some way but if you are interested feel free to contact me off-list. =Stefan -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGW1V9cVCZDfrn+pMRAt0TAJ4n0BPLDu1EBqqZg5RtIy4tEsLsJgCeJQFW yePaEzQ9FX65+SoTGxs8B6M= =TUw5 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] any codec passthru mode
Hi all, My configuration is: USER (connects to) ASTERISK---(connects to)---CARRIER-OUT i want the user preffered codec to pass thru asterisk to carrier-out. what i mean is: USER (user uses g729) ASTERISK---(asterisk should use g729 for dialing out)---CARRIER-OUT instead, this is what happens USER (user uses g729) ASTERISK---(asterisk uses g711u)---CARRIER-OUT How can i force asterisk to use user preffered codec for dialing out so that my asterisk machine saves time by no conversion USER PREFERENCE IS disallow=all allow=g729 CARRIER PREFERENCE IS allow=all Anybody who can help? -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with IAX
I am attempting to use an IAX2 channel between two Asterisk systems.
This would seem to be a normal thing to do. I actually want to trunk
traffic between the two that are in remote locations. However, I have
started with what I think is a simple configuration, which should allow
for one way calling. Attached are the pertinent parts of my
configuration files. I am attempting to place a call on 192.168.253.21
to extension 105. It seems to be routing to the IAX channel, but the
channel is being rejected by the .20 box.
Any help would be appreciated.
extensions.conf from 192.168.253.21:
;
; Create an extension, 205, for trunk
;
exten = 205,1,Dial(IAX2/tecinfo:[EMAIL PROTECTED]/[EMAIL PROTECTED])
iax.conf from 192.168.253.21:
[iax-trunk]
type=peer
username=tecinfo
secret=secret
host=192.168.253.20
---
extensions.conf from 192.168.253.20:
[iax-trunk]
exten = _205,1,Macro(voicemail,${E205})
iax.conf from 192.168.253.20:
[iax-trunk]
type=user
context=iax-trunk
username=tecinfo
secret=secret
host=192.168.253.21
-
Log from 192.168.253.21:
*CLI [May 30 09:59:01] WARNING[27827]: chan_iax2.c:6959 socket_process:
Call rejected by 192.168.253.20: No authority found
Log from 192.168.253.20
*CLI [May 30 09:59:01] NOTICE[5839]: chan_iax2.c:6754 socket_process:
Rejected connect attempt from 192.168.253.21, who was trying to reach
'[EMAIL PROTECTED]'
+
This e-mail was checked by the TecInfo Content Scanning Service for potentially
harmful content, such as viruses or Spam For more information, call
800.863.5415 or visit www.tecinfo.net
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RE: [asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!
In just about every combination of configurations I have tried (unless they were blatantly incorrect) the regular CLI say nothing (except when I tried to install AMP which gave me a permission error in the spooler). My existing config I will put below. The debug says this: - --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 66.176.193.46 : 11214 (no NAT) --- Transmitting (no NAT) to 66.176.193.46:11214 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46 From: sip:[EMAIL PROTECTED];tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 949 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 66.176.193.46:11214 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46 From: sip:[EMAIL PROTECTED];tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0 To: sip:[EMAIL PROTECTED];tag=as0d27cf25 Call-ID: [EMAIL PROTECTED] CSeq: 949 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=3721d6a7 Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) --- SIP read from 66.176.193.46:4024 --- REGISTER sip:66.109.17.92 SIP/2.0 Via: SIP/2.0/UDP 66.176.193.46:11214 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 950 REGISTER Contact: sip:66.176.193.46:11214;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER User-Agent: RTC/1.2.4949 Authorization: Digest username=UXMC, realm=asterisk, algorithm=MD5, uri=sip:66.109.17.92, nonce=3721d6a7, response=4d92865d351ad10e7f8ff0b4eabfbbe8 Event: registration Allow-Events: presence Content-Length: 0 - --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 66.176.193.46 : 11214 (no NAT) --- Transmitting (no NAT) to 66.176.193.46:11214 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46 From: sip:[EMAIL PROTECTED];tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 950 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 -- Saved useragent RTC/1.2.4949 for peer UXMC --- Transmitting (no NAT) to 66.176.193.46:11214 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46 From: sip:[EMAIL PROTECTED];tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0 To: sip:[EMAIL PROTECTED];tag=as0d27cf25 Call-ID: [EMAIL PROTECTED] CSeq: 950 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: sip:66.176.193.46:11214;expires=120 Date: Wed, 30 May 2007 15:45:39 GMT Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) --- SIP read from 66.176.193.46:4024 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 66.176.193.46:11214 Max-Forwards: 70 From: UXMC sip:[EMAIL PROTECTED];tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:66.176.193.46:11214 User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 448 v=0 o=- 0 0 IN IP4 66.176.193.46 s=session c=IN IP4 66.176.193.46 b=CT:1000 t=0 0 m=audio 32744 RTP/AVP 97 111 112 6 0 8 4 5 3 101 a=rtpmap:97 red/8000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:112 G7221/16000 a=fmtp:112 bitrate=24000 a=rtpmap:6 DVI4/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 - --- (11 headers 20 lines) --- Sending to 66.176.193.46 : 11214 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] --- Reliably Transmitting (no NAT) to 66.176.193.46:11214 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46 From: UXMC sip:[EMAIL PROTECTED];tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0 To: sip:[EMAIL PROTECTED];tag=as55eebfec Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5e7f413d Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) Found user
Re: [asterisk-users] Bottom line on fax reception
On Mon, 28 May 2007, Andrew Joakimsen wrote: If you want proper (T.38) fax support then pick up a Cisco AS53xx, AS54xx, AS58xx, 2600, 3600, 7200. You need IOS 12.1 or above, double check Cisco for specific WIC, RAM and other misc. requirements of course! ... even then, if you're running it over the Internet or even a slightly jittery WAN, it won't always work. I have a 5300 and it's been tried. Works pretty well over private IP transport, but fails about 25-50% of the time when run over any substantial segment of the public Internet. Also, at least one thing I've run into / heard suggests that the AS5300s are much better than the 5400s at handling low-speed analog and preserving data integrity. I know a CLEC that had to use 5300s in front of their fax-to-email server to get it to work properly, with the results from the 5400s they otherwise use for MGWs being very mixed. Your mileage may vary on that one, though. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct + +1-678-954-0671 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] False ring problem
No. First the carrier and then the asterisk to the user. Look at the
diagram:
First: Carrier --Ring--- Asterisk
Then: AsteriskRing- User
Rgds, Ricardo.
Rizwan Hisham escreveu:
Do you mean to say that -- first the carrier sends the msg
to us to ring and then the end user sends the msg to ring?
On 5/30/07, Ricardo Martins
[EMAIL PROTECTED] wrote:
It seems that the ring issue
is on the CARRIER-OUT signaling. It's
sending you a SIP-Ring-Message and your asterisk-box is sending it to
the callee. The second green line ".is ringing" apears jut because
your box received a ring signal from the CARRIER-OUT. Got the point?
I don't know what the "left from hold" means but seems to be related to
the situation when we push the "flash" button on the phone to put "on
hold" and flash again to put "out of hold". But I'm realy not sure
about it.
Rgds, Ricardo Martins
Rizwan Hisham escreveu:
Here is my CLI output:
Called
[EMAIL PROTECTED]
--
SIP/CARRIER-OUT-007d0310
is ringing
-- Call on
SIP/CARRIER-OUT-007d0310 left from hold
--
SIP/CARRIER-007d0310
is making progress passing it to SIP/pepsi-00f267e0
i clearly notice that when the first orange cli msg appears then the
actual ringing starts. like this tone -- tone -- totone -- tone, and if
the callee is busy then tone -- tone -- tobeep beep .
does anyone know what this means: -- Call on SIP/CARRIER-OUT-007d0310
left from hold
On 5/30/07, Rizwan Hisham
[EMAIL PROTECTED] wrote:
Maybe
its a bug in asterisk 1.4.2
On 5/30/07, Rizwan Hisham [EMAIL PROTECTED]
wrote:
There
is no R/r option in my dial application.im
only using gM option here is the dialplan:
exten=
_1X.,1,NoOp("Dialing Local!!!")
exten=
_1X.,2,Dial(Sip/[EMAIL PROTECTED],,gM(payasyougo^${CDR(accountcode)}^${CDR(userfield)}))
exten= _1X.,3,Hangup
On 5/30/07,
Ricardo Martins
[EMAIL PROTECTED] wrote:
You should (must!)
remove
any r/R parameter from your command. If you
do that, no false ring will be generated anymore...
Att, Ricardo.
Rizwan Hisham escreveu:
Hi all,
when a user dials any number, asterisk automatically generates ringing
which caller can hear, and after 2 - 3 rings asterisk detects that the
called user is busy, then caller hears busy tone. for example user
hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the
false ringing at the start so that user hears only beep beep beep if
the called user is busy. I have used the R and r options in Dial
application but they dont work.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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Rizwan Hisham
Software Engineer
AXVOICE Inc.
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Re: [asterisk-users] Help with IAX
Can you send IAX.conf of both the systems Regards, Sanjay Rajdev - Original Message - From: Malcom Kemp [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, 30 May, 2007 9:05:34 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Help with IAX ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW *
Can you cut and paste the last few relevant lines of your log file? That should help determine what is causing the core dumps. After that is determined you can file a bug report with the log file cut and paste if necessary. Is there some reason you cannot test patches on a separate test system. If it's a legitimate bug there will likely be others looking for a solution that would be willing to test. Doing things that way helps everyone including yourself. I have found the developers VERY responsive to well documented bug reports. -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Tuesday, May 29, 2007 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW * Tzafrir Cohen wrote: On Tue, May 29, 2007 at 01:06:54PM -0500, Eric ManxPower Wieling wrote: Michael Collins wrote: I think its a fair decision . 1.2 is very stable and they are not closing it all together , security issues will still be fixed . They need to concentrate more on 1.4 to make it bugfree . Fair indeed. I would guess that a completely stable 1.2 w/ security maintenance is acceptable to the majority of users. Those folks still using 1.0.x certainly aren't clamoring for new features! The great many Except that for some users 1.2.18 is NOT stable. I've had to roll back to 1.2.15 on my production servers in order to prevent core dumps at least once per day. No, I am not willing to turn my production servers into testing servers to solve this. Doing so would make me a former consultant for these customers. So basically what you're saying is that some efforts should be concentrated on 1.2 as well. So let's start with your specific problems. Are there open bugs for them? My specific problem is that Asterisk 1.2.17 and 1.2.18 (I've not tried 1.2.16) core dumps at least once per day. 1.2.15 works just fine for me. I don't know if there are open bugs. I've not opened any bugs. Any time I open a bug for a problem I have on a production server, all people want me to do is test patches to see if they fix the issue. They don't seem to understand the term production server. Sorry, but this is my JOB, not my hobby. Perhaps the customers of the developers don't care if the PBX crashes once per day, but my customers do care and I will do whatever is required to make them stop yelling at me. What made them stop yelling at me is moving back to 1.2.15. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bottom line on fax reception
I was not clear. I removed the PDF conversion only because the faxes were ot being recived correctly in most cases. The PDF conversion worked fine, but who needs a bunch of blank PDF? May as well have blank TIFF :) On 5/29/07, Doug Lytle [EMAIL PROTECTED] wrote: randulo wrote: All spam faxes arrive perfectly readable. For actual documents faxed by customers, one in 5 work. Because of this I removed the PDF Man that's awful! My experience is just the opposite. I might have a bad fax2pdf conversion once a month. I review the pdfs being archived on a regular basis. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any codec passthru mode
so you r sure you have g729 licences installed and ur * is transcoding your RTP streaming? Test the work flow with disallow=all and allow=g729, can be my mistake but I remember to read somewhere on the net any issue about codec negotiating precedence when you use allow=all. good luck On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, My configuration is: USER (connects to) ASTERISK---(connects to)---CARRIER-OUT i want the user preffered codec to pass thru asterisk to carrier-out. what i mean is: USER (user uses g729) ASTERISK---(asterisk should use g729 for dialing out)---CARRIER-OUT instead, this is what happens USER (user uses g729) ASTERISK---(asterisk uses g711u)---CARRIER-OUT How can i force asterisk to use user preffered codec for dialing out so that my asterisk machine saves time by no conversion USER PREFERENCE IS disallow=all allow=g729 CARRIER PREFERENCE IS allow=all Anybody who can help? -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help with IAX
The IAX.CONF are both the sample configs, with the addition of the two pieces that I added and posted in the email. But here they are -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sanjay Rajdev Sent: Wednesday, May 30, 2007 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Help with IAX Can you send IAX.conf of both the systems Regards, Sanjay Rajdev - Original Message - From: Malcom Kemp [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, 30 May, 2007 9:05:34 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Help with IAX ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users + This e-mail was checked by the TecInfo Content Scanning Service for potentially harmful content, such as viruses or Spam For more information, call 800.863.5415 or visit www.tecinfo.net + 192.168.253.21.iax.conf Description: 192.168.253.21.iax.conf 192.168.253.20.iax.conf Description: 192.168.253.20.iax.conf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reset Polycom phones remotely
An answer to your original question: if you can get someone _to_ the phones, on the polycom 30x's you would hold the VolDn, VolUp, DND, and Hold buttons for a while to reboot. For anyone with the 50x or 60x, you would hold the VolDn, VolUp, Messages, and Hold buttons. Moj Forum wrote: I have provisioned a bunch of Polycom 301 phones to get the config files from my ftp server. Out of the 4 phones 2 get the config file however the other 2 cannot contact the boot server. I have reboot the phones a number of times remotely (the client is 400 km away) through vnc and logging onto the web config internally. No matter what I change on the web config page it is not saved. I feel I need to reset or reformat the phones - if so how can I do this remotely? Can anyone think of a reason why these 2 phones cannot contact the boot server when the other 2 can? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial plan inquiry using GotoIf()
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan. All I basically wish to do is block a
particular caller. Sounds easy enough, but my ternary operator/plan
currently is not properly being implemented. Can anyone spot where I'm
being a momo?
All extensions get forwarded to the following macro:
[macro-forward]
; arg1 = phone number
; arg2 = timeout
; arg3 = extension (voicemail)
; arg4 = mobile number
exten = s,1,Zapateller(answer|nocallerid)
exten = s,2,PrivacyManager
exten = s,3,Wait(1)
exten = s,4,GotoIf($[${CALLERID(number)} = 15552221313]?15:5)
exten = s,5,GotoIf($[${LEN(${CALLERID(number)})} = 4]?6:8)
exten = s,6,AGI(didextlookup.agi|${CALLERID(number)})
exten = s,7,Set(CALLERID(number)=${didlookup})
exten = s,8,GotoIf($[${LEN(${CALLERID(number)})} = 10]?9:10)
exten = s,9,Set(CALLERID(number)=1${CALLERID(number)})
exten = s,10,Dial(${ARG1},${ARG2})
exten = s,11,GotoIf($[${EXISTS(${ARG4})}]?11:12)
exten = s,12,Dial(${ARG4},${ARG2})
exten = s,13,Voicemail(u${ARG3})
exten = s,14,Playback(vm-goodbye)
exten = s,15,HangUp
exten = s,105,HangUp
As you can tell, exten = s,4,GotoIf($[${CALLERID(number)} =
15552221313]?15:5) is what I recently added.
Here's what I see in the CLI logs:
-- Executing [EMAIL PROTECTED]:1] Macro(IAX2/lime-3,
forward|SIP/5609|15|5609|IAX2/[EMAIL PROTECTED]/15164766201) in new stack
-- Executing [EMAIL PROTECTED]:1] Zapateller(IAX2/lime-3,
answer|nocallerid) in new stack
-- Executing [EMAIL PROTECTED]:2] PrivacyManager(IAX2/lime-3, )
in new stack
-- CallerID Present: Skipping
-- Executing [EMAIL PROTECTED]:3] Wait(IAX2/lime-3, 1) in new stack
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/lime-3, 0?15:5) in
new stack
-- Goto (macro-forward,s,5)
It evaluates to false, hence goes to s,5. I keep dialing from that
particular number (the one in the example is clearly masked as a false
CID), and verified it's showing up as that number on callerID.
Also one last question. Say I need to add more numbers to block in the
future, is there an easier way to do this than renumbering my entire
macro? Renumbering everything is just begging for a typo which can
effectively render my dial plan broken.
Thank you kindly, everyone!
- sf
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RE: [asterisk-users] Help with IAX
In both iax.conf files change [iax-trunk] to [tecinfo] the [name] in iax.conf is what is looked for when a connection is established and you're telling it to connect with tecinfo on the username= line HTH Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan inquiry using GotoIf()
On 5/30/07, Steve Finkelstein [EMAIL PROTECTED] wrote:
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan.
snip
exten = s,4,GotoIf($[${CALLERID(number)} = 15552221313]?15:5)
It's the quotes that are messing it up... what you probably want is:
exten = s,4,GotoIf($[${CALLERID(number)} = 15552221313]?15:5)
Also, the CallerID number probably doesn't have the 1 on the front
(depending on whether or not your upstream provider sends the 1).
Also one last question. Say I need to add more numbers to block in the
future, is there an easier way to do this than renumbering my entire
macro? Renumbering everything is just begging for a typo which can
effectively render my dial plan broken.
Yes, you can use the 'n' priority, and use labels to mark the
priorities you want to jump to from your GotoIf()s.
-Jared
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Re: [asterisk-users] Dial plan inquiry using GotoIf()
You could also use the cid syntax in the extension
exten = s/ObnoxiousCallerId,1,Goto(getlost)
On 5/30/07, Steve Finkelstein [EMAIL PROTECTED] wrote:
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan. All I basically wish to do is block a
particular caller. Sounds easy enough, but my ternary operator/plan
currently is not properly being implemented. Can anyone spot where I'm
being a momo?
All extensions get forwarded to the following macro:
[macro-forward]
; arg1 = phone number
; arg2 = timeout
; arg3 = extension (voicemail)
; arg4 = mobile number
exten = s,1,Zapateller(answer|nocallerid)
exten = s,2,PrivacyManager
exten = s,3,Wait(1)
exten = s,4,GotoIf($[${CALLERID(number)} = 15552221313]?15:5)
exten = s,5,GotoIf($[${LEN(${CALLERID(number)})} = 4]?6:8)
exten = s,6,AGI(didextlookup.agi|${CALLERID(number)})
exten = s,7,Set(CALLERID(number)=${didlookup})
exten = s,8,GotoIf($[${LEN(${CALLERID(number)})} = 10]?9:10)
exten = s,9,Set(CALLERID(number)=1${CALLERID(number)})
exten = s,10,Dial(${ARG1},${ARG2})
exten = s,11,GotoIf($[${EXISTS(${ARG4})}]?11:12)
exten = s,12,Dial(${ARG4},${ARG2})
exten = s,13,Voicemail(u${ARG3})
exten = s,14,Playback(vm-goodbye)
exten = s,15,HangUp
exten = s,105,HangUp
As you can tell, exten = s,4,GotoIf($[${CALLERID(number)} =
15552221313]?15:5) is what I recently added.
Here's what I see in the CLI logs:
-- Executing [EMAIL PROTECTED]:1] Macro(IAX2/lime-3,
forward|SIP/5609|15|5609|IAX2/[EMAIL PROTECTED]/15164766201) in new stack
-- Executing [EMAIL PROTECTED]:1] Zapateller(IAX2/lime-3,
answer|nocallerid) in new stack
-- Executing [EMAIL PROTECTED]:2] PrivacyManager(IAX2/lime-3, )
in new stack
-- CallerID Present: Skipping
-- Executing [EMAIL PROTECTED]:3] Wait(IAX2/lime-3, 1) in new stack
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/lime-3, 0?15:5) in
new stack
-- Goto (macro-forward,s,5)
It evaluates to false, hence goes to s,5. I keep dialing from that
particular number (the one in the example is clearly masked as a false
CID), and verified it's showing up as that number on callerID.
Also one last question. Say I need to add more numbers to block in the
future, is there an easier way to do this than renumbering my entire
macro? Renumbering everything is just begging for a typo which can
effectively render my dial plan broken.
Thank you kindly, everyone!
- sf
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Re: [asterisk-users] Trying to dial out on teliax
Here is what I am working with now! [204] exten = 204,1,Wait() exten = 204,2,Answer exten = 204,3,Playback(demo-congrats) exten = 204,4,Hangup exten = s,1,Dial,(teliax) -- exten = s,2,Hangup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help with IAX
I did that. Got same results. I also changed the extensions on the .21 box to: exten = 205,1,Dial(IAX2/tecinfo:[EMAIL PROTECTED]/[EMAIL PROTECTED]) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Wednesday, May 30, 2007 12:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Help with IAX In both iax.conf files change [iax-trunk] to [tecinfo] the [name] in iax.conf is what is looked for when a connection is established and you're telling it to connect with tecinfo on the username= line HTH Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] + This e-mail was checked by the TecInfo Content Scanning Service for potentially harmful content, such as viruses or Spam For more information, call 800.863.5415 or visit www.tecinfo.net +___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW *
The problem with this is that if 1.2 has a bug that is making it unstable, it should be fixed to make a stable project, rather then steam rolling ahead to the next release. Further, I have seen on several occassions a security patch cause stability issues in Asterisk. On 5/30/07, Jared Smith [EMAIL PROTECTED] wrote: On 5/30/07, Steve Totaro [EMAIL PROTECTED] wrote: I do hope that when they find major security bugs like the recent SIP bug for example, that affected both 1.2.x and 1.4.x, they backport the fix. At least if the code base has not changed all that much and it is only a few lines of code. Yes, that's the whole idea of putting Asterisk 1.2 into Security Maintenance Mode (or whatever the official name is for it). Security issues will still be fixed for 1.2.x releases, but non-security-related bug fixes will only be applied to the 1.4 branch and trunk. I would anticipate that security issues will continue to be fixed until the next branch (1.6?) is released, and enough time has elapsed until 1.4 is put into security mode as well. The idea is that at any given time, you'll have: Trunk -- new features + bug fixes + security fixes Current release branch -- bug fixes + security fixes, but no new features Previous release branch -- security fixes only (after ~6 months from the date that the current release branch is released). Just to clarify, we currently have: Trunk -- new features + bug fixes + security fixes 1.4 -- bug fixes + security fixes, but no new features 1.2 -- security fixes only When 1.6 is released we'll have: Trunk -- new features + bug fixes + security fixes 1.6 -- bug fixes + security fixes, but no new features 1.4 -- bug fixes + security fixes, but no new features and about six months after 1.6 is released, we'll have: Trunk -- new features + bug fixes + security fixes 1.6 -- bug fixes + security fixes, but no new features 1.4 security fixes only The idea is to give everyone a reasonable amount of time to migrate their systems, after a new release branch is released. I think everyone realizes that a new release branch isn't automagically perfect, and it takes a little time to shake out the bugs. If you're still having problems with the 1.4 branch (or with specific versions of the 1.2 branch), I suggest you do the following to help the developers track down the problems: 1) Check to see if there are any other bug reports with the same symptoms as your own. 2) If there aren't any, fill out your own bug report. Please include as much pertinant information as possible. Does the problem only occur during high call volumes? Is it repeatable? Was a core file generated? If so, please provide a backtrace. 3) Please work with the bug marshalls and developers as they request feedback in the bug tracker. Unfortunately, we have a high number of bugs where someone reports a bug, but doesn't give any additional information when requested. 4) Try any suggested patches. I know this is difficult for some people (especially those who are running Asterisk in production systems, and don't have a test environment setup). Unfortunately, the developers can't fix the bugs without your help. -Jared ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO!
AMP does not support 1.4 and will not until AMP 2.3 is released! Bet you guys didn't think about that one! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help with IAX
I have made some changed to your config:
extensions.conf from 192.168.253.21:
;
; Create an extension, 205, for trunk
;
exten = 205,1,Dial(IAX2/ mailto:IAX2/192.168.253.20/[EMAIL PROTECTED]
tecinfo1/205)
iax.conf from 192.168.253.21:
[tecinfo1]
type=peer
username=tecinfo2
secret=secret
host=192.168.253.20
---
extensions.conf from 192.168.253.20:
[iax-trunk]
exten = _205,1,Macro(voicemail,${E205})
iax.conf from 192.168.253.20:
[tecinfo2]
type=user
context=iax-trunk
username=tecinfo
secret=secret
host=192.168.253.21
Try this and respond with error messages (if any) from both systems
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
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FW: [asterisk-users] Help with IAX
(missed one thing)
I have made some changed to your config:
extensions.conf from 192.168.253.21:
;
; Create an extension, 205, for trunk
;
exten = 205,1,Dial(IAX2/ mailto:IAX2/192.168.253.20/[EMAIL PROTECTED]
tecinfo1/205)
iax.conf from 192.168.253.21:
[tecinfo1]
type=peer
username=tecinfo2
secret=secret
host=192.168.253.20
---
extensions.conf from 192.168.253.20:
[iax-trunk]
exten = _205,1,Macro(voicemail,${E205})
iax.conf from 192.168.253.20:
[tecinfo2]
type=user
context=iax-trunk
secret=secret
host=192.168.253.21
Try this and respond with error messages (if any) from both systems
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
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Re: [asterisk-users] Problem on incoming call from Zap channel to SIP phones...
On Wed, 2007-05-30 at 08:56 -0600, Stephen Bosch wrote: Carlos Chavez wrote: On Tue, 29 May 2007 20:20:13 -0500, Eric \ManxPower\ Wieling wrote I just made another test by dialing to a Zap channel instead of a SIP phone and the call goes through without any problem. It is just when you try to dial to a SIP phone that you get the auto-congestion message. All other phones in the system are working properly, they are all registered and you can send and receive calls from anywhere except that zap channel. I'm suspicious of the Zap channel in the off-hook state. It should on-hook when on-hook and off-hook when in use. Is that channel still in off-hook? You say you made no changes, it just stopped working. Did anything *else* change around the time this problem appeared? Did someone move a device, or did you update a driver? No changes have been made in a while. The customer is very particular and very difficult to deal with so we NEVER make any changes unless it is absolutely necessary. Today I made several more tests and things are getting weirder. First I tried to connect the Vonage ATA to another port on the card just to confirm that it was not the port who was the problem. I has the exact same problem. I cannot call any SIP phone, they all give me Auto-Congestion. If I change the dialplan to dial an analog phone it goes through without a hitch, the problem only presents itself with the SIP phones. Stranger still is that I can use a SIP phone to dial to the Vonage line, only incoming calls have a problem. -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW *
Matt wrote: The problem with this is that if 1.2 has a bug that is making it unstable, it should be fixed to make a stable project, rather then steam rolling ahead to the next release. Further, I have seen on several occassions a security patch cause stability issues in Asterisk. These are my hopes as well. In addition to security related bugs, I would like to see any stability bugs quashed as well. New features, I can live without for now, but bugs affecting the stability of the product should be implemented IMO. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Need some help with IAX trunking.
I've got six systems:
AsteriskM (main)
___|
| || | |
Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5
AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other Asterisk
boxes are using ztdummy for timing, they are all using IAX trunking.
My calls come in over Sip or Zap to asteriskm and are routed to one of the
asteriskN servers based on load. The routing is done by a small AGI script
that gets the current load from a monitoring machine and then changes the
priority. Dialplan snippet:
--- Snippet ---
exten = _X.,1,AGI(manager.agi)
exten = _X.,100,Dial(IAX2/asterisk1/${EXTEN})
exten = _X.,200,Dial(IAX2/asterisk2/${EXTEN})
exten = _X.,300,Dial(IAX2/asterisk3/${EXTEN})
exten = _X.,400,Dial(IAX2/asterisk4/${EXTEN})
exten = _X.,500,Dial(IAX2/asterisk5/${EXTEN})
--- Snippet ---
This works fine for a few calls. I'm using the SIPp package to generate a
10-25 simultaneous call load. Every once in a while I starting seeing loads
of error messages on AsteriskM's console:
chan_iax2.c:7054 socket_process: Peer 'asterisk4' is now UNREACHABLE! Time:
2
chan_iax2.c:6216 socket_read: Out of idle IAX2 threads for I/O, pausing!
chan_iax2.c:909 __schedule_action: Out of idle IAX2 threads for scheduling!
chan_iax2.c:7054 socket_process: Peer 'asterisk4' is now REACHABLE! Time:
134
chan_iax2.c:6216 socket_read: Out of idle IAX2 threads for I/O, pausing!
That is just a small example, I may have 50-100 of these type of messages
scroll very quickly. If I give the system a minute everything goes back to
normal.
I would like some one who is very knowledgeable about IAX to assist me with
this problem. If someone knows a lot about IAX optimization and is willing
to work with me I would be willing to pay for their time.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
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RE: [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO!
It's not that big a deal - some gui's or third party apps will move to 1.4 some will stay on 1.2 Personally unless there is some compelling reason to move it's makes sense to stay on 1.2 until necessary or functions drive the change. As for AMP staying on 1.2 - what are the real downsides to Asterisk in that happening? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BSumrall Sent: Wednesday, 30 May 2007 2:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO! AMP does not support 1.4 and will not until AMP 2.3 is released! Bet you guys didn't think about that one! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help with IAX
From 192.168.253.20:
*CLI [May 30 14:39:03] NOTICE[6051]: chan_iax2.c:7146 socket_process:
Rejected connect attempt from 192.168.253.21, request '[EMAIL PROTECTED]'
does not exist
From 192.168.253.21:
[May 30 14:39:03] WARNING[28640]: chan_iax2.c:6959 socket_process: Call
rejected by 192.168.253.20: No such context/extension
I even changed the extension to take the pattern off:
exten = 205,1,Macro(voicemail,${E205})
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ruggles
Sent: Wednesday, May 30, 2007 1:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Help with IAX
I have made some changed to your config:
extensions.conf from 192.168.253.21:
;
; Create an extension, 205, for trunk
;
exten = 205,1,Dial(IAX2/tecinfo1/205
mailto:IAX2/192.168.253.20/[EMAIL PROTECTED] )
iax.conf from 192.168.253.21:
[tecinfo1]
type=peer
username=tecinfo2
secret=secret
host=192.168.253.20
---
extensions.conf from 192.168.253.20:
[iax-trunk]
exten = _205,1,Macro(voicemail,${E205})
iax.conf from 192.168.253.20:
[tecinfo2]
type=user
context=iax-trunk
username=tecinfo
secret=secret
host=192.168.253.21
Try this and respond with error messages (if any) from both systems
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
+
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800.863.5415 or visit www.tecinfo.net
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RE: [asterisk-users] Help with IAX
Got the same thing when I removed the username from the
192.168.253.20.iax.conf...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ruggles
Sent: Wednesday, May 30, 2007 1:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: FW: [asterisk-users] Help with IAX
(missed one thing)
I have made some changed to your config:
extensions.conf from 192.168.253.21:
;
; Create an extension, 205, for trunk
;
exten = 205,1,Dial(IAX2/tecinfo1/205
mailto:IAX2/192.168.253.20/[EMAIL PROTECTED] )
iax.conf from 192.168.253.21:
[tecinfo1]
type=peer
username=tecinfo2
secret=secret
host=192.168.253.20
---
extensions.conf from 192.168.253.20:
[iax-trunk]
exten = _205,1,Macro(voicemail,${E205})
iax.conf from 192.168.253.20:
[tecinfo2]
type=user
context=iax-trunk
secret=secret
host=192.168.253.21
Try this and respond with error messages (if any) from both systems
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
+
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Re: [asterisk-users] Problem on incoming call from Zap channel to SIP phones...
Carlos Chavez wrote: On Wed, 2007-05-30 at 08:56 -0600, Stephen Bosch wrote: Carlos Chavez wrote: On Tue, 29 May 2007 20:20:13 -0500, Eric \ManxPower\ Wieling wrote I just made another test by dialing to a Zap channel instead of a SIP phone and the call goes through without any problem. It is just when you try to dial to a SIP phone that you get the auto-congestion message. All other phones in the system are working properly, they are all registered and you can send and receive calls from anywhere except that zap channel. I'm suspicious of the Zap channel in the off-hook state. It should on-hook when on-hook and off-hook when in use. Is that channel still in off-hook? You say you made no changes, it just stopped working. Did anything *else* change around the time this problem appeared? Did someone move a device, or did you update a driver? No changes have been made in a while. The customer is very particular and very difficult to deal with so we NEVER make any changes unless it is absolutely necessary. Today I made several more tests and things are getting weirder. First I tried to connect the Vonage ATA to another port on the card just to confirm that it was not the port who was the problem. I has the exact same problem. I cannot call any SIP phone, they all give me Auto-Congestion. If I change the dialplan to dial an analog phone it goes through without a hitch, the problem only presents itself with the SIP phones. Stranger still is that I can use a SIP phone to dial to the Vonage line, only incoming calls have a problem. Sounds like you need to stop obsessing over the Zap ports. Your problem is with the SIP phones. Can two SIP phones on that system call each other? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO!
On 5/30/07, BSumrall [EMAIL PROTECTED] wrote: AMP does not support 1.4 and will not until AMP 2.3 is released! I'm sorry to hear you think our decision (I say our, as I was at the Asterisk Developers' Conference where the decision was made) will kill the AMP project. Personally, I don't think the situation is as dire as you say. I'm quite sure the AMP developers will step up to the plate and support Asterisk 1.4 in due time. When that will be I can't say, as I'm not active in the AMP community. I can't image it would take that long to move over to Asterisk 1.4, as the dialplan changes aren't *that* extensive between 1.2 and 1.4. (Obviously any code that ties into the internal C APIs of Asterisk will take longer to port.) Bet you guys didn't think about that one! Actually, we did. As a matter of fact, I was *very* vocal at the conference in stating that we needed to give users, integrators, and projects like AMP a substantial warning before putting Asterisk 1.2 in security maintenance mode, as they need time to react. At the same time, I don't think anyone should expect the Asterisk developers to base all their decisions completely on the timetables of outside projects (like AMP). There is a plethora of projects and programs out there that tie into Asterisk, and if we as developers waited for every single one to move over to Asterisk 1.4, we'd never accomplish anything. There's simply a finite set of resources (developers and bug marshalls in this case), and a decision had to be made on how best to use those resources. Personally, I think it would be great if there were more communication between the outside projects and the Asterisk developers, so that there isn't so much animosity when decisions like this are made. In short, the decision is probably going to cause some short-term discomfort for some people, but I truly believe it's a good decision for the long-term health and sanity of the Asterisk developers and Asterisk community in general. No, we're not trying to kill off AMP or any other outside project -- we're trying to make Asterisk (and by extension, anything that uses or adds on to Asterisk) as great as possible. -Jared ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW *
2007/5/30, Matt [EMAIL PROTECTED]: The problem with this is that if 1.2 has a bug that is making it unstable, it should be fixed to make a stable project, rather then steam rolling ahead to the next release. Further, I have seen on several occassions a security patch cause stability issues in Asterisk. I'm not aware of any easy way to turn an unstable server into a stable one nor aware of any bug-free application software. And if such software did exist, what happens with security patches from Operating System, or hardphone upgrades or devices you don't manage ? The real questions are : - Which open bugs are keeping you from proving given telephony services ? - Do you then have a way to lower your service level or to investigate ? - How many open serious bugs are still affecting 1.4 ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help with IAX
Well this may not feel like progress, but it is. You no longer have an
authentication issue, you now have a routing issue. Could you attach a copy
of the extension.conf file on 192.168.253.21?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Malcom Kemp
Sent: Wednesday, May 30, 2007 3:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Help with IAX
From 192.168.253.20:
*CLI [May 30 14:39:03] NOTICE[6051]: chan_iax2.c:7146 socket_process:
Rejected connect attempt from 192.168.253.21, request '[EMAIL PROTECTED]' does
not exist
From 192.168.253.21:
[May 30 14:39:03] WARNING[28640]: chan_iax2.c:6959 socket_process: Call
rejected by 192.168.253.20: No such context/extension
I even changed the extension to take the pattern off:
exten = 205,1,Macro(voicemail,${E205})
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Re: [asterisk-users] False ring problem
WRONG! WRONG! WRONG! The r option to Dial provides a fake ringback. This option tells Asterisk, no matter what sound it should be providing to the caller (busy, congestion, ringback, the number you called is disconnected, etc), it should unconditionally provide a ringing tone to the caller. Asterisk will provide ringing tone to the caller by default. Let me repeat that: Asterisk will provide a ringing tone to the caller by default. The r option is an OVERRIDE. If you do not get a ringing tone then there is something wrong. If you mask the issue by using the r option you will have trouble finding the real cause of the problem. Jaswinder Singh wrote: Do you have 'r' in ur dial command ? 'r' makes asterisk produce ring . On 30/05/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R and r options in Dial application but they dont work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] False ring problem
Rizwan Hisham wrote: Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R and r options in Dial application but they dont work. Remove the r option from Dial. I assume you have the following: SIP Phone - Asterisk w/FXO card - POTS line If you are using AMP or any other GUI for Asterisk, then my advice is not valid, since those GUIs take over everything, hide the important stuff, and add options to Dial that you never see. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple host= in sip.conf
Bryan Laird wrote: for inbound connections how does asterisk manage host=host-name returning multiple A records... will it allow authentication for any of the IP's returned? I would assume that in the case of 'inbound' if you specify a host-name that you have PTR records for you could do it in one entry again I'm making a blind assumption. As I understand it, Asterisk does a DNS lookup on load/reload and uses whatever the first IP address returned. allow= and deny= is what should be used for access control. Not the host= line. The host= line is normally used for Asterisk - Device stuff. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reset Polycom phones remotely
On 5/30/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: An answer to your original question: if you can get someone _to_ the phones, on the polycom 30x's you would hold the VolDn, VolUp, DND, and Hold buttons for a while to reboot. For anyone with the 50x or 60x, you would hold the VolDn, VolUp, Messages, and Hold buttons. Moj Moj, is this more of a hard reset to factory defaults? Does cutting the power with a power over ethernet switch do what you need? Forum wrote: I have provisioned a bunch of Polycom 301 phones to get the config files from my ftp server. Out of the 4 phones 2 get the config file however the other 2 cannot contact the boot server. I have reboot the phones a number of times remotely (the client is 400 km away) through vnc and logging onto the web config internally. No matter what I change on the web config page it is not saved. I feel I need to reset or reformat the phones - if so how can I do this remotely? Can anyone think of a reason why these 2 phones cannot contact the boot server when the other 2 can? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem on incoming call from Zap channel to SIP phones...
On Wed, 2007-05-30 at 14:49 -0500, Eric ManxPower Wieling wrote: No changes have been made in a while. The customer is very particular and very difficult to deal with so we NEVER make any changes unless it is absolutely necessary. Today I made several more tests and things are getting weirder. First I tried to connect the Vonage ATA to another port on the card just to confirm that it was not the port who was the problem. I has the exact same problem. I cannot call any SIP phone, they all give me Auto-Congestion. If I change the dialplan to dial an analog phone it goes through without a hitch, the problem only presents itself with the SIP phones. Stranger still is that I can use a SIP phone to dial to the Vonage line, only incoming calls have a problem. Sounds like you need to stop obsessing over the Zap ports. Your problem is with the SIP phones. Can two SIP phones on that system call each other? Everything else in the system works, all sip phones can call each other and the PSTN. They have a GSM adapter on the same card and they can place and receive calls. Only calls coming from the Vonage ATA have this problem. -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Hi,
Please take a look at
http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+gotchas
iax.conf The new threading model is great, but the default of 10 threads is
way too low. Symptoms include total loss of audio until the channel is hung
up.
- in general section, add: iaxthreadcount = 200
- in general section, add: iaxmaxthreadcount = 1000
Hope this helps.
Regards,
Cristian
On 5/30/07, David Ruggles [EMAIL PROTECTED] wrote:
Need some help with IAX trunking.
I've got six systems:
AsteriskM (main)
___|
| || | |
Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5
AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other Asterisk
boxes are using ztdummy for timing, they are all using IAX trunking.
My calls come in over Sip or Zap to asteriskm and are routed to one of the
asteriskN servers based on load. The routing is done by a small AGI script
that gets the current load from a monitoring machine and then changes the
priority. Dialplan snippet:
--- Snippet ---
exten = _X.,1,AGI(manager.agi)
exten = _X.,100,Dial(IAX2/asterisk1/${EXTEN})
exten = _X.,200,Dial(IAX2/asterisk2/${EXTEN})
exten = _X.,300,Dial(IAX2/asterisk3/${EXTEN})
exten = _X.,400,Dial(IAX2/asterisk4/${EXTEN})
exten = _X.,500,Dial(IAX2/asterisk5/${EXTEN})
--- Snippet ---
This works fine for a few calls. I'm using the SIPp package to generate a
10-25 simultaneous call load. Every once in a while I starting seeing
loads
of error messages on AsteriskM's console:
chan_iax2.c:7054 socket_process: Peer 'asterisk4' is now UNREACHABLE!
Time:
2
chan_iax2.c:6216 socket_read: Out of idle IAX2 threads for I/O, pausing!
chan_iax2.c:909 __schedule_action: Out of idle IAX2 threads for
scheduling!
chan_iax2.c:7054 socket_process: Peer 'asterisk4' is now REACHABLE! Time:
134
chan_iax2.c:6216 socket_read: Out of idle IAX2 threads for I/O, pausing!
That is just a small example, I may have 50-100 of these type of messages
scroll very quickly. If I give the system a minute everything goes back to
normal.
I would like some one who is very knowledgeable about IAX to assist me
with
this problem. If someone knows a lot about IAX optimization and is willing
to work with me I would be willing to pay for their time.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
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Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW *
On Wed, 2007-05-30 at 21:54 +0200, Olivier wrote: 2007/5/30, Matt [EMAIL PROTECTED]: The problem with this is that if 1.2 has a bug that is making it unstable, it should be fixed to make a stable project, rather then steam rolling ahead to the next release. Further, I have seen on several occassions a security patch cause stability issues in Asterisk. I'm not aware of any easy way to turn an unstable server into a stable one nor aware of any bug-free application software. And if such software did exist, what happens with security patches from Operating System, or hardphone upgrades or devices you don't manage ? The real questions are : - Which open bugs are keeping you from proving given telephony services ? - Do you then have a way to lower your service level or to investigate ? - How many open serious bugs are still affecting 1.4 ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Not trying to start a flame war, however the issues that I see with 1.2 and 1.4 are very similar to the issues relating to Redhat and Fedora. Redhat didn't want to continue supporting the open source model and convinced? the end user community to support all of the old releases based on the number of deployed systems. If the user community really doesn't want the versions to go away, then they won't allow it to happen. My question is this: Will digium provide the needed support to the community to allow them to continue supporting the 1.2 release, or will this prove to be related to business issues that the user community is not aware of, which will result in a much broader support of callweaver? my $.02 which probably isn't worth $.02! dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW *
On 5/30/07, Lee Jenkins [EMAIL PROTECTED] wrote: These are my hopes as well. In addition to security related bugs, I would like to see any stability bugs quashed as well. New features, I can live without for now, but bugs affecting the stability of the product should be implemented IMO. It seems there's still a lot of confusion, so I'm going to spell it out so that it's painfully obvious. The Asterisk developers don't add features to a release branch. That means that there should have been no new features in the 1.2 branch since 1.2.0 was released, and no new features in 1.4 since 1.4.0 was released. New features are only added to trunk. (Now, I can think of at least one minor exceptions to this rule, but it was just that -- a minor exception.) Now, let's do some quick math... Asterisk 1.2.0 was released in November of 2005. That means almost 18 months since the feature freeze for the Asterisk 1.2 branch. (In reality, it's longer than that because there was a feature freeze on the 1.2 branch before 1.2.0 was released.) Asterisk 1.4.0 was released in December of last year, but has been in a feature freeze state for almost a year now. -Jared ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple host= in sip.conf
On Wed, 2007-05-30 at 15:29 -0500, Eric ManxPower Wieling wrote: Bryan Laird wrote: for inbound connections how does asterisk manage host=host-name returning multiple A records... will it allow authentication for any of the IP's returned? I would assume that in the case of 'inbound' if you specify a host-name that you have PTR records for you could do it in one entry again I'm making a blind assumption. As I understand it, Asterisk does a DNS lookup on load/reload and uses whatever the first IP address returned. allow= and deny= is what should be used for access control. Not the host= line. The host= line is normally used for Asterisk - Device stuff. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Does that mean that even when dynamic dns entries exist and the time to live is set to 15 minutes asterisk will continue to try using the old expired results? Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.4 reproducibly dumps core on Solaris 10
Hi Frank, You need to replace the line 1427: handle_nodebugchan_deprecated, NULL, with handle_nodebugchan_deprecated, , and build it again. Unfortunately in Solaris a NULL field causes a SIGSEGV whenever you are going to print it out. The problem arises when calling: ast_cli(fd, %25.25s %s\n, e-_full_cmd, e-summary); I've tested in version 1.4.2 and works fine. It should work in 1.4.4 as well. Marios ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem on incoming call from Zap channel to SIP phones...
Carlos Chavez wrote: On Wed, 2007-05-30 at 14:49 -0500, Eric ManxPower Wieling wrote: Can two SIP phones on that system call each other? Everything else in the system works, all sip phones can call each other and the PSTN. They have a GSM adapter on the same card and they can place and receive calls. Only calls coming from the Vonage ATA have this problem. I missed the first part of the thread. Can you paste the CLI output of a successful call (SIP phone to SIP phone) and an unsuccessful call (Zap to SIP)? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!
You have too many codecs allowed.
disallow=all and allow=ulaw in [general] and in each of the device
sections of iax.conf. If that works, then you can start from there and
try to get the codec you really want.
BSumrall wrote:
after 18 hours, over 200 pages of reading, a complete reinstall of asterisk
I am down to this.
extensions.conf
[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1
[default]
exten = 8005181896,1,Dial,(IAX2/UXMC)
exten = s,1,Answer()
(I tried)
exten = _1XX,1,DIAL,(IAX2/teliax/${EXTEN},30,tr)
(as well)
iax.conf
[general]
port=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
register = :[EMAIL PROTECTED]
[teliax]
context=default
type=friend
host=voip-co3.teliax.com
auth=md5
user=
secret=x
disallow=all
allow=ulaw
allow=alaw
allow=gsm
sip.conf
[UXMC]
user=xxx
context=internal
type=friend
qualify=yes
nat=no
secret=
canreinvite=no
host=dynamic
nat=no
If I put back previous config, I can call into the 1800 number and here that
silly chick heckle me from my server!
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RE: [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO!
If anything this should motivate the FreePBX developers a bit more. -Original Message- From: Jared Smith [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 30, 2007 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO! On 5/30/07, BSumrall [EMAIL PROTECTED] wrote: AMP does not support 1.4 and will not until AMP 2.3 is released! I'm sorry to hear you think our decision (I say our, as I was at the Asterisk Developers' Conference where the decision was made) will kill the AMP project. Personally, I don't think the situation is as dire as you say. I'm quite sure the AMP developers will step up to the plate and support Asterisk 1.4 in due time. When that will be I can't say, as I'm not active in the AMP community. I can't image it would take that long to move over to Asterisk 1.4, as the dialplan changes aren't *that* extensive between 1.2 and 1.4. (Obviously any code that ties into the internal C APIs of Asterisk will take longer to port.) Bet you guys didn't think about that one! Actually, we did. As a matter of fact, I was *very* vocal at the conference in stating that we needed to give users, integrators, and projects like AMP a substantial warning before putting Asterisk 1.2 in security maintenance mode, as they need time to react. At the same time, I don't think anyone should expect the Asterisk developers to base all their decisions completely on the timetables of outside projects (like AMP). There is a plethora of projects and programs out there that tie into Asterisk, and if we as developers waited for every single one to move over to Asterisk 1.4, we'd never accomplish anything. There's simply a finite set of resources (developers and bug marshalls in this case), and a decision had to be made on how best to use those resources. Personally, I think it would be great if there were more communication between the outside projects and the Asterisk developers, so that there isn't so much animosity when decisions like this are made. In short, the decision is probably going to cause some short-term discomfort for some people, but I truly believe it's a good decision for the long-term health and sanity of the Asterisk developers and Asterisk community in general. No, we're not trying to kill off AMP or any other outside project -- we're trying to make Asterisk (and by extension, anything that uses or adds on to Asterisk) as great as possible. -Jared ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] (no subject)
Thanks; I have made the change and I will try it tomorrow! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cristian N. Bradiceanu Sent: Wednesday, May 30, 2007 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) Hi, Please take a look at http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+g otchas iax.conf The new threading model is great, but the default of 10 threads is way too low. Symptoms include total loss of audio until the channel is hung up. * in general section, add: iaxthreadcount = 200 * in general section, add: iaxmaxthreadcount = 1000 Hope this helps. Regards, Cristian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agents.conf from realtime static (Solved)
On Tue, 2007-05-29 at 17:13 -0400, Jared Smith wrote: On 5/29/07, Carlos Chavez [EMAIL PROTECTED] wrote: I am using Asterisk 1.4.4 on a CentOS 5 machine for a small call center with 6 agents. I am using realtime for queues and sip and I am also trying to use realtime static to load agents.conf. The only problem I am having is that no agents are loaded when I start Asterisk. I have to manually do a module reload chan_agent.so so the agents get loaded from the database. This sounds like it *might* be a problem with the order in which your modules are being loaded. To fix this, you might want to manually add the following lines to your modules.conf file, to make sure that chan_local and chan_sip get loaded *before* chan_agent: load = chan_local.so load = chan_sip.so load = chan_agent.so Give that a try and let me know if it works for you. What needs to be preloaded is the realtime engine for Mysql. So you just have to insert the following line into /etc/asterisk/modules.conf: preload = res_config_mysql.so -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple host= in sip.conf
David Boyd wrote: Does that mean that even when dynamic dns entries exist and the time to live is set to 15 minutes asterisk will continue to try using the old expired results? You would have to try it and see. I do not know all the DNS oddities of Asterisk. Asterisk's DNS support is the worst I have ever seen in any piece of software that supports DNS. Unfortunately fixing this is no easy task. Search the mailing list archives for more information. I don't have DNS issues with Asterisk because I never give Asterisk anything except an IP address. Devices that change IP addresses register to Asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple host= in sip.conf
On Wed, 2007-05-30 at 18:03 -0500, Eric ManxPower Wieling wrote: David Boyd wrote: Does that mean that even when dynamic dns entries exist and the time to live is set to 15 minutes asterisk will continue to try using the old expired results? I can also say that my experience in putting DynDNS hostnames in sip.conf do not even get mapped to IP addresses at all. I have ALWAYS had to put an actual IP for it to not grab it from eth0 by default. It never errors out while reading the config file, or logs anything - I just know it never looked up the IP for me. I have not personally tried 1.4 yet, but I would (like you) wish it to look it up and create the appropriate headers instead of me relying on my firewall to re-write them. -Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW *
Jared Smith wrote: Now, let's do some quick math... Asterisk 1.2.0 was released in November of 2005. That means almost 18 months since the feature freeze for the Asterisk 1.2 branch. (In reality, it's longer than that because there was a feature freeze on the 1.2 branch before 1.2.0 was released.) Asterisk 1.4.0 was released in December of last year, but has been in a feature freeze state for almost a year now. And Asterisk 1.2.18 STILL has show stopping bugs. This does not make me feel all warm and fuzzy about moving to 1.4.x. In fact, the idea of moving to 1.4.x right now scares the hell out of me. I don't like crashing PBXs. I don't like users screaming at me because they lost a million dollar contract because their phones were down half the day. Asterisk is a PBX. It should not have to be upgraded as often as some Microsoft server. One of my customers are looking at moving to a 4 year upgrade cycle (mostly because that is the max length of support from the distro vendor they use) No matter how much you test before deployment there will be issues that are not seen until you put the system under significant load in a real usage situation. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [1.2.18] Wrong steps in extensions.conf?
On Tue, 29 May 2007 07:39:40 -0400, in
gmane.comp.telephony.pbx.asterisk.user Luis Morales wrote:
# send the result over callerid ;-)
$AGI-exec('SetCallerId', $response-content);
$AGI-exec('Dial', $ext);
$AGI-hangup();
I'm sorry, but I don't understand why you added this in the script
that updates the web page.
Isn't LookupCIDName blocking, ie. the next step won't be run until
LookupCIDName is done?
exten = group,1,LookupCIDName
exten = group,n,AGI(web.agi|${CALLERID(num)}|${CALLERID(name)})
exten = group,n,Dial(${EXT204})
BTW, is LookupCIDName a binary program, or a script somewhere?
Thank you.
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[asterisk-users] Re: OpenVox A400P01on thin client?
On Tue, 29 May 2007 10:23:18 -0300, in gmane.comp.telephony.pbx.asterisk.user Gustavo Cordeiro wrote: No, but I think that you can't install this OpenVox board in this NetStation case, because the card is a full length PCI and the PC case supports only half length PCI cards. Thanks guys for the feedback. I'll check what kind of PCI cards those small form-factor PCs handle. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan inquiry using GotoIf()
Thanks for the help on this thread all.
It would make sense if I write an AGI and incorporate a DB backend to
check against numbers I want explicitly dropped. If anyone has such a
utility, I'd love to -not- reinvent the wheel. Otherwise, I'll go whip
it up and probably provide a web frontend for adding/removing numbers.
- sf
C F wrote:
It fails because the right function is ${CALLERID(num)}
On 5/30/07, Steve Finkelstein [EMAIL PROTECTED] wrote:
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan. All I basically wish to do is block a
particular caller. Sounds easy enough, but my ternary operator/plan
currently is not properly being implemented. Can anyone spot where I'm
being a momo?
All extensions get forwarded to the following macro:
[macro-forward]
; arg1 = phone number
; arg2 = timeout
; arg3 = extension (voicemail)
; arg4 = mobile number
exten = s,1,Zapateller(answer|nocallerid)
exten = s,2,PrivacyManager
exten = s,3,Wait(1)
exten = s,4,GotoIf($[${CALLERID(number)} = 15552221313]?15:5)
exten = s,5,GotoIf($[${LEN(${CALLERID(number)})} = 4]?6:8)
exten = s,6,AGI(didextlookup.agi|${CALLERID(number)})
exten = s,7,Set(CALLERID(number)=${didlookup})
exten = s,8,GotoIf($[${LEN(${CALLERID(number)})} = 10]?9:10)
exten = s,9,Set(CALLERID(number)=1${CALLERID(number)})
exten = s,10,Dial(${ARG1},${ARG2})
exten = s,11,GotoIf($[${EXISTS(${ARG4})}]?11:12)
exten = s,12,Dial(${ARG4},${ARG2})
exten = s,13,Voicemail(u${ARG3})
exten = s,14,Playback(vm-goodbye)
exten = s,15,HangUp
exten = s,105,HangUp
As you can tell, exten = s,4,GotoIf($[${CALLERID(number)} =
15552221313]?15:5) is what I recently added.
Here's what I see in the CLI logs:
-- Executing [EMAIL PROTECTED]:1] Macro(IAX2/lime-3,
forward|SIP/5609|15|5609|IAX2/[EMAIL PROTECTED]/15164766201) in new stack
-- Executing [EMAIL PROTECTED]:1] Zapateller(IAX2/lime-3,
answer|nocallerid) in new stack
-- Executing [EMAIL PROTECTED]:2] PrivacyManager(IAX2/lime-3, )
in new stack
-- CallerID Present: Skipping
-- Executing [EMAIL PROTECTED]:3] Wait(IAX2/lime-3, 1) in new
stack
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/lime-3, 0?15:5) in
new stack
-- Goto (macro-forward,s,5)
It evaluates to false, hence goes to s,5. I keep dialing from that
particular number (the one in the example is clearly masked as a false
CID), and verified it's showing up as that number on callerID.
Also one last question. Say I need to add more numbers to block in the
future, is there an easier way to do this than renumbering my entire
macro? Renumbering everything is just begging for a typo which can
effectively render my dial plan broken.
Thank you kindly, everyone!
- sf
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Re: [asterisk-users] Re: [1.2.18] Wrong steps in extensions.conf?
Hi Vincent,
To rewrite caller id with the result of your query
$AGI-exec('SetCallerId', $response-content);
This is only example. You can remove this line from the script.
Remember call the script with this parameters:
exten = group,n,AGI(web.agi|${CALLERID(num)}|${CALLERID(name)}|
${EXT204})
Regards,
Luis Morales
On Thu, 2007-05-31 at 01:48 +0200, Vincent wrote:
On Tue, 29 May 2007 07:39:40 -0400, in
gmane.comp.telephony.pbx.asterisk.user Luis Morales wrote:
# send the result over callerid ;-)
$AGI-exec('SetCallerId', $response-content);
$AGI-exec('Dial', $ext);
$AGI-hangup();
I'm sorry, but I don't understand why you added this in the script
that updates the web page.
Isn't LookupCIDName blocking, ie. the next step won't be run until
LookupCIDName is done?
exten = group,1,LookupCIDName
exten = group,n,AGI(web.agi|${CALLERID(num)}|${CALLERID(name)})
exten = group,n,Dial(${EXT204})
BTW, is LookupCIDName a binary program, or a script somewhere?
Thank you.
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[asterisk-users] Delays on E1 Delivered via SHDSL
I have an Asterisk system with a TE110P installed and connected to an ISDN E1 PRI that is delivered via a 2mb SHDSL connection. I am experiencing delays (the type of delay you would get on an international call) during calls. I am wondering if anyone could advise, would the problem be with any part of the Asterisk system or is the problem with the fact that the ISDN is delivered over the internet? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] montavista and Asterisk
Does anybody have any experience with Asterisk on Montavista Linux? Cheers, Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Delays on E1 Delivered via SHDSL
I doubt it's the PRI itself SHDSL isn't part of the internet per se, its just an access technology. SHDSL is just synchronous DSL which can be used to deliver E1s over. ISDN PRI's are delivered in a 2Mbit/sec G703/G704 frame and will give you lots of alarms if they are having any issues It could be your toll provider at the end of it is routing calls in ways that cause delays, but less likely to be the PRI Cheers duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Sent: Thursday, 31 May 2007 12:18 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Delays on E1 Delivered via SHDSL I have an Asterisk system with a TE110P installed and connected to an ISDN E1 PRI that is delivered via a 2mb SHDSL connection. I am experiencing delays (the type of delay you would get on an international call) during calls. I am wondering if anyone could advise, would the problem be with any part of the Asterisk system or is the problem with the fact that the ISDN is delivered over the internet? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Application Developer
I'm looking to hear from any application developers in Argentina specifically or other South/Central American countries. Please understand this isn't a dial plan or remote installation I'm looking for but an actual application developer. If this fits your description please email me details on; Size of company (number of full time/versus contract staff) Location Previous asterisk applications you have developed. And please include any specific experience in AMI and Presence. You'll need at least 2 project references I can contact and a reasonable grasp of English as my Spanish sucks - sorry. Probably looking for about 2-3 developers for a 4 week project. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Application Developer
Dean, You of all people should know that posting job listings to the user's list is abuse of the list. Question: How does your post have anything to do with the use of Asterisk? Answer: It doesn't. Question: How does your post have anything to do with Business and Asterisk? Answer: It does. Question: What do the above two questions and answers indicate? Answer: You are abusing the User's list with a post that clearly belongs on the Biz list. Thanks, Steve Totaro http://www.asteriskhelpdesk.com http://www.asteriskhelpdesk.com/component/option,com_wrapper/Itemid,37/ KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Wednesday, May 30, 2007 10:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Application Developer I'm looking to hear from any application developers in Argentina specifically or other South/Central American countries. Please understand this isn't a dial plan or remote installation I'm looking for but an actual application developer. If this fits your description please email me details on; Size of company (number of full time/versus contract staff) Location Previous asterisk applications you have developed. And please include any specific experience in AMI and Presence. You'll need at least 2 project references I can contact and a reasonable grasp of English as my Spanish sucks - sorry. Probably looking for about 2-3 developers for a 4 week project. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Application Developer
Dude chill - I actually tried the biz list first but for some reason it kept holding it back saying implicit destination - no idea based on the text what the issue is. As for asteriskhelpdesk dont worry I'm getting to it - I'll be posting the RFQ on your job site by this time tomorrow. Hadn't forgotten about it :P Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, 30 May 2007 11:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Application Developer Dean, You of all people should know that posting job listings to the user's list is abuse of the list. Question: How does your post have anything to do with the use of Asterisk? Answer: It doesn't. Question: How does your post have anything to do with Business and Asterisk? Answer: It does. Question: What do the above two questions and answers indicate? Answer: You are abusing the User's list with a post that clearly belongs on the Biz list. Thanks, Steve Totaro http://www.asteriskhelpdesk.com http://www.asteriskhelpdesk.com/component/option,com_wrapper/Itemid,37/ KB3OPB From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Wednesday, May 30, 2007 10:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Application Developer I'm looking to hear from any application developers in Argentina specifically or other South/Central American countries. Please understand this isn't a dial plan or remote installation I'm looking for but an actual application developer. If this fits your description please email me details on; Size of company (number of full time/versus contract staff) Location Previous asterisk applications you have developed. And please include any specific experience in AMI and Presence. You'll need at least 2 project references I can contact and a reasonable grasp of English as my Spanish sucks - sorry. Probably looking for about 2-3 developers for a 4 week project. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to use sable (festival) markup with asterisk
Hi, I want to use festival with asterisk to play a text with sable tags. have some body any idea about it Nasir Iqbal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
