Re: [asterisk-users] WAV file best sound quality
On 6/14/07, Matt [EMAIL PROTECTED] wrote: Ahh I didn't see that in the first post. Yes Mr. SpamSucks is correct. You should use 8khz @ 16bits. Using 8khz @ 8bits will sound like a drowning goat under water. I am a specialist in low bit width audio: My TRS-80 was able to input audio for samlping through its cassette port, so 1-bit audio was born. Handy for decoding Morse code or RTTY, no good for vmail or digital audio music. :) ___ --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 hang (Critical Response)
HI all, Recently, I got the following message from CLI and finally the asterisk will hang. Anyone can tell me how to fix the problem or why it will happen. Thanks. Jun 17 14:18:02 DEBUG[24573] channel.c: Avoiding initial deadlock for 'SIP/1127-008d65f0' Jun 17 14:22:45 ERROR[24696]: chan_sip.c:11337 sipsock_read: We could NOT get the channel lock for SIP/1589-0087cdd0! Jun 17 14:22:45 ERROR[24696]: chan_sip.c:11338 sipsock_read: SIP MESSAGE JUST IGNORED: CANCEL Jun 17 14:22:45 ERROR[24696]: chan_sip.c:11339 sipsock_read: BAD! BAD! BAD! Jun 17 14:28:04 WARNING[25368]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 103 (Critical Response) ___ --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 GUI
Tzafrir Cohen wrote: On Sat, Jun 16, 2007 at 08:55:24PM +0100, Senad Jordanovic wrote: Brett Crapser wrote: On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote: Paul Hales wrote: GUI bad! CLI good! PaulH Really...? So explain why every major PBX manufacturer has GUI of some sort? Surely they would have had CLI only if GUI is bad!!! Senad Senad - it is really to cover the inability of 'average' people to understand CLI. CLI is useful for small/simple dial tone installations. Anything above that even very competent administrator will make syntax/logical errors. Hence automation is required. Automation does not imply GUI. Bad GUIs get in the way of automation. Automation is another subject/scope. However, GUI is collection of knowledge and experience. If applied correctly it can only improve the company offerings. I have personally spent years learning CLI in order to apply it to initial design of our GUI- PBXware. Thousands installation after, I have no full knowledge of CLI any more and I do not need to. It is embedded into PBXware and our team has collective knowledge of the whole solution. That is something CLI can NOT offer since detailed knowledge/training is required individually from the vary basics. That translates into: GUI - team/company knowledge, less training, faster time to market CLI - knowledge of individual / unnecessary dependency/training /longer time to market Senad ___ --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] VPN on Asterisk
On Sunday 17 June 2007 08:25:23 am [EMAIL PROTECTED] wrote: Hi, Greetings to All, Im looking for some help on configuring VPN on the Asterisk PBX that I have hosted in US. Im currently in Middle East and as everyone knows some countries here has taboo to VOIP. Im not able to get phy phones registered to my PBX as they are blocking SIP and IAX2. Hence im looking for a VPN solution. For this first i need to setup VPN on my server .. Am i right? Well if anyone has experience in the whole setup how to make it run, a guide would be much appreciated with some pointer to equipment that are wel suited for the setup. Thanks in advance. Danny ___ --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 I'm in Middle East also and don't have problems with SIP and IAX2:) Try OpenVPN. It's easy to setup and has many features. -- Dominik Zalewski | System Administrator OpenCraft t- +2 02 336 0003 w- http://www.open-craft.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
Re: [asterisk-users] VPN on Asterisk
On 17 Jun 2007, at 06:25, [EMAIL PROTECTED] wrote: Hi, Greetings to All, Im looking for some help on configuring VPN on the Asterisk PBX that I have hosted in US. Im currently in Middle East and as everyone knows some countries here has taboo to VOIP. Im not able to get phy phones registered to my PBX as they are blocking SIP and IAX2. Hence im looking for a VPN solution. For this first i need to setup VPN on my server .. Am i right? Well if anyone has experience in the whole setup how to make it run, a guide would be much appreciated with some pointer to equipment that are wel suited for the setup. Thanks in advance. Danny Danny, I'd be very interested to hear which countries are blocking IAX2. If they are just blocking it by port number, you can always configure it to run on a different port. If they are blocking it by content inspection (unlikely but possible I suppose) you could try turning on encryption. Personally I'd try doing both the above before going down the VPN route. However I know folks have VPNs working very well. Tim. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] Mitel 5340 IP Phone
Hi all, Just a quick query; has anyone on here tried the Mitel 5340 IP Phone with Asterisk? If so, how did you find it - any problems, missing features etc? I've had a Google around and the general consensus seems good - I actually have a phone on its way to me early this week, but just thought I'd start investigating now ;) We have a live Asterisk 1.2 server, and an Asterisk 1.4 server currently setup for testing. Regards, Andy Neillans Systems Designer Blueberry Consultants Ltd ___ --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 GUI
On Sun, Jun 17, 2007 at 08:22:23AM +0100, Senad Jordanovic wrote: Tzafrir Cohen wrote: On Sat, Jun 16, 2007 at 08:55:24PM +0100, Senad Jordanovic wrote: Brett Crapser wrote: On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote: Paul Hales wrote: GUI bad! CLI good! PaulH Really...? So explain why every major PBX manufacturer has GUI of some sort? Surely they would have had CLI only if GUI is bad!!! Senad Senad - it is really to cover the inability of 'average' people to understand CLI. CLI is useful for small/simple dial tone installations. Anything above that even very competent administrator will make syntax/logical errors. Hence automation is required. Automation does not imply GUI. Bad GUIs get in the way of automation. Indeed incorrect phrasing. The GUI is not directly related to that. A bad system may include a GUI that is very convinient for a small number of objects, but get in the way of applying unexpected types of changes. Another typical situation is that a system is built with a certain flow in mind, and that flow is not good enough for all the cases. And often makes many simple tasks complicated. Automation is another subject/scope. However, GUI is collection of knowledge and experience. If applied correctly it can only improve the company offerings. I have personally spent years learning CLI in order to apply it to initial design of our GUI- PBXware. Thousands installation after, I have no full knowledge of CLI any more and I do not need to. It is embedded into PBXware and our team has collective knowledge of the whole solution. That is something CLI can NOT offer since detailed knowledge/training is required individually from the vary basics. That translates into: GUI - team/company knowledge, less training, faster time to market CLI - knowledge of individual / unnecessary dependency/training /longer time to market Actually, you need specific training of the specific system, as well as ability to debug generic Asterisk problems. Not to mention the poor souls who need to support a varity of systems. For them those systems just add complexity. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 Faxing
Any one able to assist, Please Paradise Dove wrote: so how to avoid CPC?? On 6/14/07, C F [EMAIL PROTECTED] wrote: Its called CPC On 6/12/07, Kyle Vorster [EMAIL PROTECTED] wrote: Hello, Sorry if this is a real dumb question but when sending a fax and the end user does not enable fax on their side and then just hangs up does not force asterisk to end the call. So it keeps the trunk open until its killed by a Flash Operator. Please assist if any one understands me. Kind Regards, Kyle Virster ___ --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 ___ --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] MixMonitor Problem
Asif Raza wrote: exten = sss-NOANSWER,1,VoiceMail([EMAIL PROTECTED]) exten = sss-NOANSWER,2,Goto(salesivr,s,4) I want monitor to be disabled on priority s,3. Can someone please point out what I am doing wrong here. You can't have it disabled on priority 3, you need to do the following: exten = sss-NOANSWER,1,StopMonitor() exten = sss-NOANSWER,n,VoiceMail([EMAIL PROTECTED]) exten = sss-NOANSWER,n,Goto(salesivr,s,4) 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] Transfer caller direct to voicemail
If you are trying to send a call to a users voicemail why don't you just do
this ?
exten = _3XX,1,Voicemail(${EXTEN:[EMAIL PROTECTED])
(This is assuming that you have 2 digit VM box's and you want to transfer
the call to 3+VM number).
- Original Message -
From: Drew Gibson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, June 15, 2007 9:01 PM
Subject: Re: [asterisk-users] Transfer caller direct to voicemail
Hi Wes,
thanks for the suggestion but I have gone a simpler route suggested by
Leonardo Kamache with
exten = _*[1-3]XX,1,Wait(1)
exten = _*[1-3]XX,n,Voicemail(${EXTEN:[EMAIL PROTECTED]|u)
exten = _*[1-3]XX,n,Hangup()
I had assumed the * would have been eaten by features in features.conf
but there is nothing configured there to use *!
regards,
Drew
Wes Baehr wrote:
Drew,
I've written a tiny patch that duplicates the functionality of a blind
transfer but sets a variable (VMXFER) before transferring. The dialplan
simply looks for the VMXFER variable, and if found, will direct the call
directly to voicemail.
This way, instead of making the operators learn they have to push
ADDITIONAL
digits to complete the transfer, they just use a different # sequence. (I
have mine set up as #8ext -- 8 for V)
Email me directly and I can make up a patch for you (for 1.4 anyway).
Wes Baehr
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Drew Gibson
Sent: Tuesday, June 12, 2007 11:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Transfer caller direct to voicemail
Hi,
Our operator frequently gets requests to transfer a call directly to
voicemail in order for the caller to leave a message without disturbing
the callee. Basicly, I'm looking for a blindxfer to vm.
My first thought was to prepend a digit (eg 7) to the extension but this
does not fit well with our dialplan.
According to an article on voip-info.org [EMAIL PROTECTED] appears to
implement this as #*XXX. I assume they are using an application map in
features.conf but I cannot see a way to pass the required extension to
the VoiceMail() application.
Can this be done in features.conf?
regards,
Drew
--
Drew Gibson
Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com
--
Drew Gibson
Systems Administrator
OANDA Corporation
www.oanda.com
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[asterisk-users] Upgrade cisco SIP phone 7940
Hi All, My current 7940 phones use P0S3-06-3-00. I'd like to upgrade them so they're not massively out of date. I found a page at http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx that gives some info, and using the cisco links there have tried to upgrade. According to the procedures, I should be able to upgrade, but once the phones loaded and reboots it says it downgrades again and reboots, then the cycle starts again. Anyone had any success in doing this? Adrian Marsh 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] MixMonitor Problem
On 6/16/07, Asif Raza [EMAIL PROTECTED] wrote:
Hi,
I am facing some issues while using MixMonitor and StopMonitor. My
extensions logic is attached below:
exten = s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b)
exten = s,2,Dial(SIP/101,13)
exten = s,3,StopMonitor()
exten = s,4,NoOp(Dial Status: ${DIALSTATUS})
exten = s,5,Goto(sss-${DIALSTATUS},1)
exten = sss-NOANSWER,1,VoiceMail([EMAIL PROTECTED])
exten = sss-NOANSWER,2,Goto(salesivr,s,4)
As evident from the dialplan I only want to record the call when
Dial(SIP/101,13) is successful.
After that I disable recording by issuing the StopMonitor command. Now
the problem is that when the status of dial is NOANSWER the voicemail
recording is also recorded and saved.
It is only after I hangup that I see the following print on the console
End MixMonitor Recording SIP/192.168.0.10.172-081c67c0
I want monitor to be disabled on priority s,3. Can someone please
point out what I am doing wrong here.
Ok, i was thinking this over several times without any clue. Now i
finally remembered - MixMonitor didn't natively had StopMonitor
support. StopMonitor() is for Monitor().
So, you have to 1st check do you have StopMixMonitor(). If not, you
can try a patch from here http://bugs.digium.com/view.php?id=6122
Or use regular Monitor(). I'm still using it on 1.2 without mixing, as
i had some stability issues with MixMonitor(). For now i prefer to mix
recordings in nightly cron job.
Regards,
Atis
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[asterisk-users] SIP Peering--call terminated prematurely
I am attempting to establish SIP peering between Asterisk and an AltiGen soft PBX. This is my first experience with SIP peering. I can successfully make both inbound and outbound calls to/from a softphone on the AltiGen system (network access is provided by a PRI on the Asterisk system), but they are disconnected unexpectedly. The attachment is a redirect of the Asterisk CLI during a call that is disconnected prematurely. Here's what's in SIP.conf: [altigen] type=friend username=altigen secret=coolbeans host=dynamic deny=0.0.0.0/0.0.0.0 permit=10.0.2.150/255.255.255.255 qualify=yes disallow=all allow=ulaw context=altigen-inbound dtmfmode=rfc2833 The machines are a couple feet apart on a LAN through a 100MB switch. I'd appreciate any help. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax dklog Description: Binary data ___ --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] Upgrade cisco SIP phone 7940
On Sun, 2007-06-17 at 13:45 +0100, Adrian Marsh wrote: According to the procedures, I should be able to upgrade, but once the phones loaded and reboots it says it downgrades again and reboots, then the cycle starts again. Try disabling all the tftp boxes on the cisco IP Phone except the tftp used to upgrade to SIP. Maybe you have any other tftp config that downloads another firmware. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] VPN on Asterisk
On 08:25, Sun 17 Jun 07, [EMAIL PROTECTED] wrote: Hi, Greetings to All, Im looking for some help on configuring VPN on the Asterisk PBX that I have hosted in US. Im currently in Middle East and as everyone knows some countries here has taboo to VOIP. Im not able to get phy phones registered to my PBX as they are blocking SIP and IAX2. Hence im looking for a VPN solution. For this first i need to setup VPN on my server .. Am i right? Well if anyone has experience in the whole setup how to make it run, a guide would be much appreciated with some pointer to equipment that are wel suited for the setup. Have a look at OpenVPN. Real easy to setup and it works fine. http://www.openvpn.net -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] Upgrade cisco SIP phone 7940
Be sure your OS79XX.TXT and SIPDefault.cnf file and SIP[MACADDRESS].cnf file all agree on the version of software the phones are to be running. For example OS79XX.TXT should read: P0S3-08-2-00, and in SIPDefault.cnf a line should read: image_version:P0S3-08-2-00. If you were trying to run P003-08-2-00 The image_version line is not required in SIP[MACADDRESS].cnf file so I wouldn't put it there otherwise if you have many phones you'll have to edit each phone each time you change the software. kw _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Marsh Sent: Sunday, June 17, 2007 8:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Upgrade cisco SIP phone 7940 Hi All, My current 7940 phones use P0S3-06-3-00. I'd like to upgrade them so they're not massively out of date. I found a page at http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx that gives some info, and using the cisco links there have tried to upgrade. According to the procedures, I should be able to upgrade, but once the phones loaded and reboots it says it downgrades again and reboots, then the cycle starts again. Anyone had any success in doing this? Adrian Marsh ___ --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] Upgrade cisco SIP phone 7940
Thanks Guillermo, I figured it out. I'd missed the config in SIPDefault.cnf Another question though - some of my users have used Cisco Callmanager before, and are used to being able to provision their own Address books via PC (sync to outlook etc). Anyone know hows this is done? Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M. Sent: 17 June 2007 15:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Upgrade cisco SIP phone 7940 On Sun, 2007-06-17 at 13:45 +0100, Adrian Marsh wrote: According to the procedures, I should be able to upgrade, but once the phones loaded and reboots it says it downgrades again and reboots, then the cycle starts again. Try disabling all the tftp boxes on the cisco IP Phone except the tftp used to upgrade to SIP. Maybe you have any other tftp config that downloads another firmware. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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 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
[asterisk-users] SIP Peering--call terminated prematurely
I am attempting to establish SIP peering between Asterisk and an AltiGen soft PBX. This is my first experience with SIP peering. I can successfully make both inbound and outbound calls to/from a softphone on the AltiGen system (network access is provided by a PRI on the Asterisk system), but they are disconnected unexpectedly. The attachment is a redirect of the Asterisk CLI during a call that is disconnected prematurely. Here's what's in SIP.conf: [altigen] type=friend username=altigen secret=coolbeans host=dynamic deny=0.0.0.0/0.0.0.0 permit=10.0.2.150/255.255.255.255 qualify=yes disallow=all allow=ulaw context=altigen-inbound dtmfmode=rfc2833 The machines are a couple feet apart on a LAN through a 100MB switch. I'd appreciate any help. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax dklog Description: Binary data ___ --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] SIP Peering--call terminated prematurely
I am attempting to establish SIP peering between Asterisk and an AltiGen soft PBX. This is my first experience with SIP peering. I can successfully make both inbound and outbound calls to/from a softphone on the AltiGen system (network access is provided by a PRI on the Asterisk system), but they are disconnected unexpectedly. The attachment is a redirect of the Asterisk CLI during a call that is disconnected prematurely. Here's what's in SIP.conf: [altigen] type=friend username=altigen secret=coolbeans host=dynamic deny=0.0.0.0/0.0.0.0 permit=10.0.2.150/255.255.255.255 qualify=yes disallow=all allow=ulaw context=altigen-inbound dtmfmode=rfc2833 The machines are a couple feet apart on a LAN through a 100MB switch. I'd appreciate any help. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax ___ --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 Peering--call terminated prematurely
Please do not post same thing again and again . It wont help you get better replies , Post you asterisk cli output while call is in progress and when it disconnects prematurely . On 18/06/07, Don Kelly [EMAIL PROTECTED] wrote: I am attempting to establish SIP peering between Asterisk and an AltiGen soft PBX. This is my first experience with SIP peering. I can successfully make both inbound and outbound calls to/from a softphone on the AltiGen system (network access is provided by a PRI on the Asterisk system), but they are disconnected unexpectedly. The attachment is a redirect of the Asterisk CLI during a call that is disconnected prematurely. Here's what's in SIP.conf: [altigen] type=friend username=altigen secret=coolbeans host=dynamic deny=0.0.0.0/0.0.0.0 permit=10.0.2.150/255.255.255.255 qualify=yes disallow=all allow=ulaw context=altigen-inbound dtmfmode=rfc2833 The machines are a couple feet apart on a LAN through a 100MB switch. I'd appreciate any help. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax ___ --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] CNAM.
So, is there anyone out there that provides rather generic but comprehensive CNAM-style directory services via SIP, to end-users? So I can put names to my calling numbers? Thanks! -- 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] CNAM.
I signed up for www.got-name.com about a week or two ago... seems to work fine with Asterisk, so long as you use 1.2 or 1.4 (doesn't work at all with 1.0). Good pricing, no minimums, no monthly fees, no setup fees. I originally saw them mentioned on one of these asterisk lists... either biz or users... so I gave them a try. Since I'm using 1.0 code on my production box, I can't use it right now... but I tested it on 1.2 and it worked good... so when I finally get my production machine replaced in a month or two I'll start using it full time. So I can't really comment on how well it works under pressure right now... just onesy-twosy seems to work fine. -- Nick On Sun, 17 Jun 2007, Alex Balashov wrote: So, is there anyone out there that provides rather generic but comprehensive CNAM-style directory services via SIP, to end-users? So I can put names to my calling numbers? Thanks! -- 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 ___ --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] CNAM.
Thanks Nick. Do they charge per directory dip, or in some other unit? -- 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] CNAM.
Yes... 1.5 cents per dip... you prepay the fees... and they deduct from the prepaid amount. You can start with $5.00 which seems like a low-risk to check it out at least. The CLEC I use is more expensive that that for CNAM, and they want to do it on EVERY incoming call, even wrong numbers, whether it's answered or not, per PRI. So since I get several thousand wrong numbers a month, and only 100 or so calls that I actually CARE what the CNAM is on those calls, I can set it up in Asterisk to only do the dip for certain DNIS numbers. I calculated that instead of $70+/month this will cost me $1.50/month. Nice savings. :-) I just hope it's reliable when the call volume picks up more. -- Nick On Sun, 17 Jun 2007, Alex Balashov wrote: Thanks Nick. Do they charge per directory dip, or in some other unit? -- 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 ___ --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 GUI
On Sat, 2007-06-16 at 17:10 -0400, Dean Collins wrote: Brett, The demand for asterisk GUI's could be that the world primarily consists of four year olds (as you put it - I call them customers) and not geeks with pocket protectors and Vi skills to tame all tasks. When you realize that IP Telephony/Asterisk was restricted to such a small band of users when it was pure coding with Vi and .conf files and now with GUI's like Trixbox you have a much wider base of users experimenting and implementing. Of course that's not to say that Trixbox is the be-all and end-all..personally I think that there is a hell of a lot missing (/wrong) with the Trixbox/Fonality product and a lot that could be/should be done differently/better. but that's for another email. I would have to agree with you - it's not that GUI's are bad as such, the main issue I have is that people ask me ho to do this and that, and I gowell, we are going to have to remove your GUI PaulH ___ --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] Regarding call transfer feature
Hello all, I'm testing the Asterisk 1.4.2 version under X86 and ppc platform. While implementing the call transfer feature in X86 platform, I cud able to transfer the call from one extension to other. But when i tried to transfer it for the second time, it gets failed even the call establishment between the two extensions is not taking place. Then I need to quit the xlite phones and register it again to test the transfer for the second time. In PPC platform i've tested it, but it worked for 2 or 3 times, but now it fails at each time. Anybody facing this problem in ur platform. Kindly advice regarding this issue. Thanks Regards, B.Priyalatha ___ --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
