Re: [Asterisk-Users] Forwarded call flag
On Tue, 8 Mar 2005, Tom Samplonius wrote: On Tue, 8 Mar 2005 13:36:39 -0700, Dr. Matthew Roller [EMAIL PROTECTED] wrote: When I forward my PSTN phone(Qwest) to my cellphone and someone calls it, my cellphone(ATT) shows an arrow next to the caller id showing it is a forwarded call, is there any way to set that forward flag when forwarding a phone call through asterisk? It is possible using PRI signalling (ni2 for sure). When running q931 debugging on a PRI gateway, I can see that incoming calls that have been forwarded actually have the redirecting number listed, and the type of redirection (always, na/busy, etc). I really doubt that libpri has the capability to build these kinds of messages, let alone an API to set the fields. And then try to find a PRI provider won't strip the messages (called GTD, if you talk to them). There are several possible fields in isdn for this (isn't isdn great? ;-). Some of these can be set in libpri, some can not. * using Call Deflection on the isdn link before answering. Possible with bristuffed versions of Asterisk. * using Explicit Call Transfer for an answered call. Same as above. * possibly using Redirecting Number, if your telco allows that from the user to the network. * Using the special arrangement option of ETS 300 092-1 paragraph 9.4 and Annex B. This allows the sending of two Calling Party Number elements, one of which can be set arbitrarily to the caller id of the forwarded call. This seems to be available all over Europe at least. Support is being added to Asterisk by Frank Sautter. * Using ROSE invokes for Diverting Leg Information. Already in libpri. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Another Newbie Question
Jim Van Meggelen [EMAIL PROTECTED] writes: You could do that with two tin cans and a string! ;-P ...so, the next time you want to complain about your phone service, why don't you try using two Dixie cups and a string? We don't care. We don't have to. We're the Phone Company. --Lily Tomlin as Ernestine on Saturday Night Live, 1976. -tih -- Don't ascribe to stupidity what can be adequately explained by ignorance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this response
(on call out). Any suggestions? I don't think it is a problem with the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my understanding.
Free world does work for calling out however. So I know at least that
works.
-- Got SIP response 400 Bad request back from 147.135.0.128
Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
to authenticate on INVITE to 'PP
sip:[EMAIL PROTECTED];tag=as5b80cade'
On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
First off... please cancel previous amplification request. I have
implemented your ideas with the same errored result.
I am not sure that we are not making it thru authentication. From my
digging and comparing packet dumps comparing the soft phone to asterisk
they have identical transactions through the ACK reply (the last one
on the debug below). The softphone seems to be authenticated after the
ACK. I am a newbie to debugging this stuff. I just want to get it
working.
Thanks everyone in advance for your help. I am certainly very very
happy to try anything.
Based on Luki's suggestions I...
Changed sip.conf...
[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=zjh018g8f8
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
Changed extensions.conf...
exten = _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice
for 30 seconds
exten = _8X.,2, congestion() ; No answer, nothing
exten = _8X., 102, busy() ;
End result...
Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
to authenticate on INVITE to '6050
sip:[EMAIL PROTECTED];tag=as545ccba3'
SIP debug...
-- Executing Dial(SIP/6050-132b,
SIP/[EMAIL PROTECTED]|30) in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205
v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
(no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username=6050,realm=asterisk,nonce=42d82e9b,uri=sip:
[EMAIL PROTECTED],algorithm=MD5,response=420e39b35648a10c
129dd4fb5f97ec47
Contact: 6050 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED];tag=as2f065f18
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 64.4.192.110:5060
com*CLI
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
6 headers, 0 lines
com*CLI
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED];tag=SD38rq699-
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
WWW-Authenticate: DIGEST
realm=BroadWorks,algorithm=MD5,nonce=1110353299563
Content-Length: 0
8 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
meaningful subject [was: Re: [Asterisk-Users] Another Newbie Question]
On Wed, Mar 09, 2005 at 06:38:24PM +1100, Callum McGillivray wrote: Hey all, Hi, welcome to this list My apologies if this sounds blindingly obvious, but am I correct in saying that I can use Asterisk to connect two extensions and make calls between them without needing an actual telephone line at all ? I figure it's possible. As I said, probably blindingly obvious. but my techies have gone home for the evening and I was looking for an answer before I left. Suppose someone will have the same question a year from now. He'll try to do the Right Thing and search the archives of this list first. He may get some hits for his search from this thread, but will dismiss them, because the title of the thread was a newbie question and gives no hint to the fact that we're talking about connecting extensions. Cheers -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Help Site - cut down on Mailing List questions
On Wed, Mar 09, 2005 at 02:47:29PM +1100, Mike Sander wrote: This is a re-post as it was pointed out that I replied to a different thread instead of creating a new post. Sorry for the additional traffic. Mike Dear All, I understand the excitement surrounding a service like Asterisk, and how easy it is to jump in and ask a heap of questions. I also know how frustrating it can be dealing with a 200+ post per day mailing list as one of the question answerers. When I discovered Asterisk, I had a lot of study to do, because there are no real-world examples out there, just the trivial ones on the tiki and in the manual. I hope to propose a solution. I have (in a small time) downloaded and set up a repositor where we should all post our conf files, in an effort to get a big resource of a lot of different setups that we know just work. The program is simple, and looks like crap and is a testiment to my programming skills (or lack thereof). If anyone feels like re-coding or hosting this, let me know. You can find this at: asterconf.hopto.org (i think this has popups for the free DNS) or home.exetel.com.au/azyc/asterconf The wiki has a section of exammple setups and configurations. What is the atvantage of your separate site? Please take that as constructive critisism. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT Far End Traversal
Leo Ann Boon wrote: Another question... Are you aware of a SIP ATA or phone that has some kind of VPN (i.e. PPTP) client embedded in? This would make the NAT problem go away nicely and provide added security... The Zulty's phones support VPN. Then again, many firewalls don't pass through VPN traffic nicely. Would be cool if we can have a phone that supports SSL VPNs like OpenVPN. Agreed. In my experience, OpenVPN is a breeze to work with. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please help with install *
Hi On Tue, Mar 08, 2005 at 11:53:14AM -0800, Victoria Alexandru wrote: [snip] Checking out from CVS: [EMAIL PROTECTED] victoria]# cd /usr/src [EMAIL PROTECTED] src]# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot [EMAIL PROTECTED] src]# cvs login Logging in to :pserver:[EMAIL PROTECTED]:2401/usr/cvsroot CVS password: [EMAIL PROTECTED] src]# cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds ... U asterisk-sounds/sounds/wx/temperature.gsm U asterisk-sounds/sounds/wx/wind-chill.gsm U asterisk-sounds/sounds/wx/winds.gsm [EMAIL PROTECTED] src]# One general priciple: RPM is for reproducable builds. When you build packages as root you don't get reproducable builds: the %install script can easily install on the real system. It may seem longer, but the result is a reproducable build. The general rule of thumb is that you build everything as a user, even the kernel packages. I don't know if Mandrake have kernel module packages. SuSE seem to have some. It basically only requires some convensions as to where the files will be. In Debian the zaptel-source package puts its files under /usr/src/module/zaptel and I currently manually copy those files to the build tree to generate the kernel-spcific zaptel-modules package. Just in case you don't have rpm configured to build packages as your user, I wrote a script a couple of years ago to create that configuration: http://iglu.org.il/~tzafrir/mkrpmconf -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please help with install * SOLVED
On Tue, 2005-03-08 at 14:28 -0800, Victoria Alexandru wrote: I'm not registered with wiki, but I can tell what was the mod: In rhconfig.h, in line 43 you'll find . I'll try to email Mandrake people to have certitude but for now what I did was to remove one pair of . I believe this is a typo, unless is something missing between . Thats why I say I need to signal this to mandrake and have a confirmation. Things like this are probably why there are many problems with RH and MDK mentioned on this list. (rhconfig.h points to RH?). The first thing I always do is download and configure a plain vanilla kernel from kernel.org. In the past I've found it impossible to recompile MDK kernels from their source using their original .config. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: astcc - how to use **
hi all asterisk user can you help me to find the way for hangup any call by pressing any key like ** or ## in astcc and place another call without providing calling card number. bashir i search google to find out a - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, March 09, 2005 12:45 AM Subject: meaningful subject [was: Re: [Asterisk-Users] Another Newbie Question] On Wed, Mar 09, 2005 at 06:38:24PM +1100, Callum McGillivray wrote: Hey all, Hi, welcome to this list My apologies if this sounds blindingly obvious, but am I correct in saying that I can use Asterisk to connect two extensions and make calls between them without needing an actual telephone line at all ? I figure it's possible. As I said, probably blindingly obvious. but my techies have gone home for the evening and I was looking for an answer before I left. Suppose someone will have the same question a year from now. He'll try to do the Right Thing and search the archives of this list first. He may get some hits for his search from this thread, but will dismiss them, because the title of the thread was a newbie question and gives no hint to the fact that we're talking about connecting extensions. Cheers -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Subject: Re: [Asterisk-Users] What combination of pwlib and openh323 are required to get Asterisk-oh323 v0.7.1 to compile
Thanks Vamsi I have not been able to locate pwlib - 1.6.6 or openh323 1_13_5. I found the latest versions through sourceforge and I found some older versions on another site, but not these versions. This has been quite frustrating. Anyway, I think by using the asterisk-oh323 branch under channels in the asterisk source tree I will have more luck. At present it seems to compile successfully, but fails linking due to a lib expat, which I have no idea where that comes from. Regards Mark Date: Tue, 8 Mar 2005 13:18:30 +0530 From: Vamsi Pottangi [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] What combination of pwlib and openh323 are required to get Asterisk-oh323 v0.7.1 to compile To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII This worked for me on Fedora Core 3 pwlib - 1_6_6 openh323 - 1_13_5 asterisk-oh323 - 0.7.1 cp pwlib-v1_6_6-src.tar.gz openh323-v1_13_5-src.tar.gz asterisk-oh323-0.7.1.tar.gz /usr/src/ cd /usr/src tar zxf pwlib-v1_6_6-src.tar.gz tar zxf openh323-v1_13_5-src.tar.gz tar zxf asterisk-oh323-0.7.1.tar.gz - Set Environment variables PWLIBDIR=/usr/src/pwlib OPENH323DIR=/usr/src/openh323 LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib cd /usr/src/pwlib ./configure make opt cd /usr/src/openh323 ./configure -- Remove the line 433 (:protected) in /usr/src/openh323/include/gkserver.h else you would get the below error during compilation /usr/src/openh323/include/gkserver.h:434: error: `virtual H323Transaction::Response H323GatekeeperRRQ::OnHandlePDU()' is protected -- make opt cd /usr/src/asterisk-oh323-0.7.1 Edit makefile and set the paths/options according to your system. Type make to build the oh323wrap library and the ASTERISK OH323 channel driver. Type make install to install the binaries. This will also install a sample configuration file, if there isn't one. Hope this of help to you Cheers, ~Vamsi On Tue, 8 Mar 2005 11:41:19 +0800, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi there I have Asterisk running beautifully on our test server. Over the past few days I have been tearing my hair out trying to compile various versions of asterisk-oh323 on various versions of pwlib and openh323. pwlib is now up to 1.8.3 and openh323 is now 1.15.2 stable. asterisk-oh323 is currently 0.7.1 I have tried these three with many errors. I have tried 0.7.1 with pwlib 1.5.2 and openh 1.12.2 with no luck. I have tried asterisk-oh323 1.5.10 with pwlib 1.5.2 and openh323 and I still get errors. From the mailing list I have gleaned that this version of asterisk-openh323 won't work with the latest asterisk anyway, yet the readme in asterisk-oh323 says to use this version with the aforementioned versions of pwlib and openh323. I can't find the versions of pwlib and openh323 recommended in the asterisk-oh323-0.7.1 readme. The pwlib and openh323 projects always build without error. Asterisk built without errors and most everythings else. I am running a very basic Fedora Core 2 installation. What I would like to know is what is the recommended known good combination to use of asterisk-oh323, pwlib and oh323. Once I have a combination that should work, I can then ask more intelligent questions on how to get it to build properly if I still have errors. Help greatly appreciated. Regards Mark Dutton ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail - No Audio Output!
Hi all, I am able to receive voicemail in my mail box but when I try to play the audio file attachment, I hear nothing at all (yet the caller on the other end does leave a voicemail message)! Anyone had a similar problem before? Ideas are welcome! Note: I am using [EMAIL PROTECTED] 0.6 Thanks in advance, -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF out to Cell Phone
On Tue, 2005-03-08 at 14:16 -0500, John Fullington wrote: I set up a monitoring system that calls my techs when a problem occurs on one of our networks, everything works fine unless asterisk calls a cell phone in which case the tech can not respond using dtmf. It works fine if the tech call in but not if asterisk call a tech's cell phone. Anyone one have any suggestions? The application sounds interesting. Any chance you can email more about what you are actually doing? (code?) It sounds like your problem has nothing to do with mismatching Codec's or how the DTMF is being sent... etc... I have an Asterisk installation with BRI and with a premicell attached to an analogue interface (Premicell=fixed cell phone with analogue 2-wire interface that gives dial tone - like a trunk line) Perhaps I can then confirm your problem - or help with a solution? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asteriks@home
I am newest to this group and would appreciate your help! Is it possible to use quicknet phone jack with [EMAIL PROTECTED] ver 0.6? Little has been mentioned about use of quicknet products' adaptability with [EMAIL PROTECTED] I do have a couple of old jacks to startup right away. Your guide is most welcome. Thanks, Mike __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] i am missing something!
thanks for replying but no change at all any other tips,suggestions thanks in advance On Wed, 9 Mar 2005 01:44:41 -0600, Jay Milk [EMAIL PROTECTED] wrote: You'll need canreinvite=no to each sip section in sip.conf, if you want * to stay in the loop. -Original Message- From: Adnan Ahmed [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 09, 2005 1:14 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] i am missing something! Hello ppl, At initial level i configure asterisk woth only soft phones ,in which one at windows machine and other is linux i am using windows messenger and linphone respectively both phones registered with asterisk respectively problem is that they bypass asterisk on call when i send request from linphone to messenger request shown on messenger but on asterisk console nothing to and also if i send request from messenger to linphone it doesn't recognized at all my config are: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to sip-h323 using asterisk-oh323-0.7.1
hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323-sip by using asterisk as gateway. help required on sip-h323. kamran __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Music on hold
Is it true music on hold isnt supported in IAX/2? I check the docs and it doesnt show a configuration setting in IAX.conf and when I put someone on hold they dont hear the music and * doesnt start the music on hold. If it doesnt is there a way to make this work?___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call through. with 2xT1 .configuration
Hello all, It 's dificult to explain; The system I need is an box option (based on *), that I would add to an existing PABX (ie: Nortel with 600 ext). I need two E1/T2 card to plug the system between Telco (FT) and PABX (Nortel)! One card for France Telecom Side (E1a) and one other to Nortel Side (E1b). - --- Telco FT |(E1a)--|System X|(E1b)-|PABX Nortel|--600 ext. - --- The existing (Nortel) PABX must run like before the system box is plug, without any modification on it. The System box intercept only incomming call on special DID/SDA ie: 4000. Then a prompt ask the extension to be reach, and the system box call this extension via the E1b link through the Nortel PABX. All other incoming and all outgoing call pass through the system box, transparency. Just a small system is requiered, I think, because there is only 30 simultaneous calls. No VOIP, No voicemail, No SIPphone, NO Extension. The goal is to unsucribed average 200 DID from France Telecom with this system. I search some explains or samples to configure * to do that. Regards Florent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Should ICMP port unreachable generate a BYE request?
Hi all, I'm researching random call drops on our Asterisk and would like to make sure whether it's something wrong with our VoIP provider or with the Asterisk. I sniffed traffic between Asterisk and our VoIP provider's SIP gateway, and observed that in the middle of the conversation an RTP stream originating from Asterisk gets an ICMP port unreachable from provider's SIP gateway at random times and conversation seems to go on for a while, but after a while a few more port unreachables are observed and Asterisk sends BYE request to both parties. I wonder if it is it normal for Asterisk to send BYE requests to both parties once it gets a port unreach even though noone from the either end has hanged up the call? If yes, then why doesn't it send BYE request on the first unreach it sees? Or is it some tunable parameter that can be set via configuration files? Or should I mail the sniffer dump to my provider and ask them to fix their gateway? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Help Site - cut down on Mailing List questions
On March 9, 2005 03:49 am, Tzafrir Cohen wrote: The wiki has a section of exammple setups and configurations. What is the atvantage of your separate site? Please take that as constructive critisism. The wiki is very messy and hard to find information. And I say this as an experienced Asterisk user (multiple PRI setups, voicemail, spandsp, hard/softphones, manager interface, etc.) -- the wiki's a good idea but it is very... confusing? congested? I think that this site has some good potential. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Should ICMP port unreachable generate a BYE request?
On Wed, Mar 09, 2005 at 01:51:09PM +0200, Dipole Moment wrote: I'm researching random call drops on our Asterisk and would like to make sure whether it's something wrong with our VoIP provider or with the Asterisk. I sniffed traffic between Asterisk and our VoIP provider's SIP gateway, and observed that in the middle of the conversation an RTP stream originating from Asterisk gets an ICMP port unreachable from provider's SIP gateway at random times and conversation seems to go on for a while, but after a while a few more port unreachables are observed and Asterisk sends BYE request to both parties. I wonder if it is it normal for Asterisk to send BYE requests to both parties once it gets a port unreach even though noone from the either end has hanged up the call? If yes, then why doesn't it send BYE request on the first unreach it sees? Or is it some tunable parameter that can be set via configuration files? Or should I mail the sniffer dump to my provider and ask them to fix their gateway? You'd have to trace the code to work it out properly. But ICMP packets aren't generally passed to userspace. What's more likely is that the kernel, upon receiving sufficient of these errors, decides the connection is dead and notifies asterisk. Although, with UDP (in Linux anyway) the error can be passed back. Strange problem though, how can only some packets generate Port Unreachable, but not all. Random routing problem? Have a nice day, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Regarding Incoming Calls on PRI
Hello, I am trying to make a call from our PABX to Asterisk on PRI interface. How can iconfigure Asterisk to enter the overlap receiving state if the complete number is not obtained in setup message. Looking forward to any help in this regard Regards Nauman Bin Ali__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 Phantom Ringing
1.4.1 over here. Jerry wrote: Never had any of my 100 or so act like that. What version of code are you running? I think 1.4.1 is the latest. On Mar 8, 2005, at 8:05 PM, Ben Ruset wrote: Hello list: I have a very odd problem. Seemingly randomly, my Polycom IP600 phones will ring without a call being placed to it. That is to say, a random phone will ring. Nothing shows up under Caller ID. Even the buttons that light up to show an incoming call do not light up. If you pick up the handset, you can hear the phone ring through the speaker. Hanging up the phone makes it stop ringing. Then, sometime later, it will happen on another random extension. Is this a common problem? Where can I look to start diagnosing this? Thanks! -ben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cicso 7912 phones 3 out of 8 not grabbing thegkMAC file
On Tue, 08 Mar 2005 17:34:31 -0500, Jerry Geis [EMAIL PROTECTED] wrote: SEPDEFAULT.cnf is not required nor recommended. The 7912 only uses the gkMAC file and the software version CP7912XXX file The gk file must be lower case.. This phone 192.168.255.250 is requesting SEPXXX It is still running a SCCP Imagege not SIP and needs Upgrading ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which box?
I'm sure this is a stupid question, but I'm not finding an answer anywhere. Do I need a dedicated box to run asterisk, or can I put in my server (running Fedora) and leverage some of the free cpu cycles and disk space? Thanks, Dunc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regarding Incoming Calls on PRI
On March 9, 2005 07:26 am, n a wrote: How can i configure Asterisk to enter the overlap receiving state if the complete number is not obtained in setup message. I take it the overlapdial=yes option isn't doing what you want? Perhaps a more detailed explanation of what you're after would help, including the output of pri debug span x with the relevant bits exposed. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which box?
Dunc, Depends on the environment you run it in. If this is main telephony system for a business, then a dedicated machine is highly desirable, and you may also want to think about redundancy and failover. If it's for your own personal use, or it's a development machine, then it can co-exist with other software with no problems. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ asterisk wrote: I'm sure this is a stupid question, but I'm not finding an answer anywhere. Do I need a dedicated box to run asterisk, or can I put in my server (running Fedora) and leverage some of the free cpu cycles and disk space? Thanks, Dunc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which box?
Separate box is best. If you are only new to asterisk go and download [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/ It's a iso you can download that does all of the configuring and setup for you automatically. Cheers dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Wednesday, March 09, 2005 7:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which box? I'm sure this is a stupid question, but I'm not finding an answer anywhere. Do I need a dedicated box to run asterisk, or can I put in my server (running Fedora) and leverage some of the free cpu cycles and disk space? Thanks, Dunc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which box?
On March 9, 2005 07:31 am, asterisk wrote: I'm sure this is a stupid question, but I'm not finding an answer anywhere. Do I need a dedicated box to run asterisk, or can I put in my server (running Fedora) and leverage some of the free cpu cycles and disk space? Thanks, That's a very open-ended question. Your accurate answer: It depends. At home I have my P3/700 (which is also an NFS/SMB server and NFS root for my mythtv box) also running asterisk. Hell the TDM430P and ethernet share the same interrupt and it works *fine* for me. This is most certainly NOT an optimal situation for most people though. At the office I have two dedicated Asterisk boxes and they work well too. For a beginner such as yourself I would *strongly* recommend using dedicated hardware until you understand enough about the interactions to be able to intelligently guess what's going to happen. It's not like you need a lot of horsepower for a regular asterisk box. I was using a P90 without MMX for a while, but it was only one FXS and one FXO port. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which box?
Just out of interest, has anyone tried Asterisk @Home on User Mode Linux? Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ dean collins wrote: Separate box is best. If you are only new to asterisk go and download [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/ It's a iso you can download that does all of the configuring and setup for you automatically. Cheers dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Wednesday, March 09, 2005 7:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which box? I'm sure this is a stupid question, but I'm not finding an answer anywhere. Do I need a dedicated box to run asterisk, or can I put in my server (running Fedora) and leverage some of the free cpu cycles and disk space? Thanks, Dunc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF out to Cell Phone
Hi John, You didn't say what kind of cellular system. If its an AMPS system (I don't think any other analogue cellular stiff exists) DTMF is quite troublesome. If it is a digital network the DTMF actually comes from the basestation, rather than the phone. Its is normally very high quality. However, its timing is nothing like the timing of the button pushes on the handset. The basestation stretches the digits to rather long ones. Possibly as much as a second each. Regards, Steve John Fullington wrote: I set up a monitoring system that calls my techs when a problem occurs on one of our networks, everything works fine unless asterisk calls a cell phone in which case the tech can not respond using dtmf. It works fine if the tech call in but not if asterisk call a tech's cell phone. Anyone one have any suggestions? Thanks John Fullington ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which box?
Why would you want to, it's a single iso, takes only 15 minutes to install make your config changes for your particular machine and then use the backup feature. You bust anything irreparable, just load the iso again and load the backup. Up and running again in under 20 mins. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alistair Cunningham Sent: Wednesday, March 09, 2005 7:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Which box? Just out of interest, has anyone tried Asterisk @Home on User Mode Linux? Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ dean collins wrote: Separate box is best. If you are only new to asterisk go and download [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/ It's a iso you can download that does all of the configuring and setup for you automatically. Cheers dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Wednesday, March 09, 2005 7:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which box? I'm sure this is a stupid question, but I'm not finding an answer anywhere. Do I need a dedicated box to run asterisk, or can I put in my server (running Fedora) and leverage some of the free cpu cycles and disk space? Thanks, Dunc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming Fax Service question
Dunno if I did make myself clear. I want to route an incoming ISDN call using the excess digits dialed. Need this for Fax. If I understood your post and the wiki, using NV(Fax|Background)Detect should Just Work, like in the example. Has anybody done this with an ISDN line? Will, if the user just dials the fax number through (which is actually 2 or 3 digits longer than the real ISDN number), the excess digits be there as DTMF? Justin Newman schrieb: If you need to dial additional digits after pickup, use the D(...) command with Dial. Why not just send the call to another extension or DID? To detect fax on the line, you can use NVFaxDetect or NVBackgroundDetect. More information on the Tikiwiki. http://www.voip-info.org/tiki-index.php?page=NVFaxDetect http://www.voip-info.org/tiki-index.php?page=NVBackgroundDetect Justin -- Best Regards, Met vriendelijke groeten, Mit freundlichen Grüßen, Timm Gebhart Personal Office B.V. Tel: +31 77 320 2923 Fax: +31 77 320 2921 [EMAIL PROTECTED] Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo for first 15 to 20 seconds
I am using asterisk with a handful of DM04B cards. Everything seems fine except for an echo on all calls on the local end of the call. In almost all cases the echo goes away after 15 to 20 seconds. I am attributing the echo going away to the echo cancellation code that was enabled when the following options are set: echocancel=yes ;echocancelwhenbridged=yesechotraining=yes ; Is there anyway short of playing with these options which I have already done to improve this echo issue. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo for first 15 to 20 seconds
I am using asterisk with a handful of DM04B cards. Everything seems fine except for an echo on all calls on the local end of the call. In almost all cases the echo goes away after 15 to 20 seconds. I am attributing the echo going away to the echo cancellation code that was enabled when the following options are set: echocancel=yes ; echocancelwhenbridged=yes echotraining=yes ; Instead of echotraining=yes, use echotraining=800 and don't forget to 'stop' and restart asterisk. A simple reload won't cut it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which box?
Dean, - It's something new and fun to try! - For testing clustering, advanced routing between Asterisks, etc, without having to buy lots of machines. - I'm not suggesting it at the minute as it's not proven, but perhaps at some point in the future, offering multiple customers their own dedicated Asterisk installation, all on the same machine. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ dean collins wrote: Why would you want to, it's a single iso, takes only 15 minutes to install make your config changes for your particular machine and then use the backup feature. You bust anything irreparable, just load the iso again and load the backup. Up and running again in under 20 mins. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alistair Cunningham Sent: Wednesday, March 09, 2005 7:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Which box? Just out of interest, has anyone tried Asterisk @Home on User Mode Linux? Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ dean collins wrote: Separate box is best. If you are only new to asterisk go and download [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/ It's a iso you can download that does all of the configuring and setup for you automatically. Cheers dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Wednesday, March 09, 2005 7:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which box? I'm sure this is a stupid question, but I'm not finding an answer anywhere. Do I need a dedicated box to run asterisk, or can I put in my server (running Fedora) and leverage some of the free cpu cycles and disk space? Thanks, Dunc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P slow getting line tone
Hello all, I just installed a TDM400P with 2 FXO modules on my asterisk server. The card works perfectly. To get users to ring out from my SIP phones i setup an extension with 0 that basically does something like this: extension = 0,1,Dial(ZAP/g1) where g1 is the group of the two FXO channels extension = 0,2,Hangup This works exactly as i want so users basically can dial 0, wait for the dialtone and then dial the requested number. The only problem that i have is that from when a user dial 0 to when i get the dialtone from the telephone line, something like 5 seconds pass... is it possible to pull this wait time down to about 1 second? or even less?? I already set in zapata.conf the immediate=yes property.. Can someone help me out? Best Regards, Fabrizio Mazzoni Macron Srl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk System() call error
I have a linux (bash) script file which is invoked via:
exten = s,3,System(./BuildMsg.sh ${SFile} msg02 msg99)
When I start asterisk with the command: asterisk -gc
This script executes as expected ('asterisk -gc' and 'asterisk -vg' also
work). However, when I try to start asterisk with the command:
'asterisk -g' the script does not execute and I get the following error
message in the 'messages' log file:
Mar 9 08:06:55 WARNING[790]: Unable to execute './BuildMsg.sh 1-1 msg02
msg99'
The script file is located in /etc/asterisk, and I have confirmed that
asterisk is looking for the script file in this location: I tried
exten = s,3,System(pwd location.out) and location.out contained
'/etc/asterisk'.
Asterisk is running as root and the group/owner of Buildmsg.sh are set to
'root' (I have also done a 'chmod 777 Buildmsg.sh' just to be sure). I am
running: Asterisk CVS-HEAD-02/17/05-11:17:10, on a linux box with GNU bash,
version 2.05b.0(1)
Any ideas as to what would be causing this behaviour are greatly
appreciated!
Jonathan
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[Asterisk-Users] Zyxel P2000W - CallerId
Caller Name set using SetCIDName or SetCallerID is not displayed by Zyxel P2000W (Firmware VWJ000F). The same problem has been mentionend before, but I did not find any solution or hint. http://lists.digium.com/pipermail/asterisk-users/2005-January/082801.html -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NuFone + VoIPJet = busy busy busy
Hi List, I'm using VoIPJet and NuFone as a fallback, and it seems that both of them are circuit busy! Also it seems that VoIPJet takes forever to return 'circuit busy' while NuFone does it instantly. At any rate, is there like a reliable third VoIP provider I can use for fallback when the two others are busy? Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice latest changes and still not working-An Additional Server
CHris, I had the exact same problem with the exact same error. My password was entered incorrectly in context section. The register line had the correct password. That is why you get incoming calls. and not outgoing. Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo for first 15 to 20 seconds
I thought echotraining=400 was the default? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, March 09, 2005 8:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Echo for first 15 to 20 seconds I am using asterisk with a handful of DM04B cards. Everything seems fine except for an echo on all calls on the local end of the call. In almost all cases the echo goes away after 15 to 20 seconds. I am attributing the echo going away to the echo cancellation code that was enabled when the following options are set: echocancel=yes ; echocancelwhenbridged=yes echotraining=yes ; Instead of echotraining=yes, use echotraining=800 and don't forget to 'stop' and restart asterisk. A simple reload won't cut it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cicso 7912 phones 3 out of 8 not grabbingthegkMAC file
Jason, You are correct. The phone is brand new and running SCCP. The tftp server has the upgrade info - the gkMAC file. the 3 phones are not picking it up. THe other 5 phones did it just fine. You are correct the phone needs upgrading. Th gkMAC file pointes to the upgrade file. The 7912 is not grabbing it. Jerry On Tue, 08 Mar 2005 17:34:31 -0500, Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / SEPDEFAULT.cnf is not required nor recommended. The 7912 only uses the gkMAC file // and the software version CP7912XXX file // // The gk file must be lower case.. / This phone 192.168.255.250 is requesting SEPXXX It is still running a SCCP Imagege not SIP and needs Upgrading ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone + VoIPJet = busy busy busy
the following is on voipjet's site: Please note we are having a temporary glitch with our New York location. Please send traffic to our West Coast Premium Server until the problem is fixed sometime today. New SERVER IP: 69.25.60.30 although i guess an email to this effect would have been nice. -yair On Wed, 09 Mar 2005 17:41:07 +0400, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Hi List, I'm using VoIPJet and NuFone as a fallback, and it seems that both of them are circuit busy! Also it seems that VoIPJet takes forever to return 'circuit busy' while NuFone does it instantly. At any rate, is there like a reliable third VoIP provider I can use for fallback when the two others are busy? Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Try changing the extension from Broadvoice1 to the actual phone number
(and don't send your secret in a public email or maybe that's Chris'):
[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=XXX
username=8475100139
Zanzamar Majere wrote:
I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this response
(on call out). Any suggestions? I don't think it is a problem with the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my understanding.
Free world does work for calling out however. So I know at least that
works.
-- Got SIP response 400 Bad request back from 147.135.0.128
Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
to authenticate on INVITE to 'PP
sip:[EMAIL PROTECTED];tag=as5b80cade'
On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
First off... please cancel previous amplification request. I have
implemented your ideas with the same errored result.
I am not sure that we are not making it thru authentication. From my
digging and comparing packet dumps comparing the soft phone to asterisk
they have identical transactions through the ACK reply (the last one
on the debug below). The softphone seems to be authenticated after the
ACK. I am a newbie to debugging this stuff. I just want to get it
working.
Thanks everyone in advance for your help. I am certainly very very
happy to try anything.
Based on Luki's suggestions I...
Changed sip.conf...
[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=DELETED
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
Changed extensions.conf...
exten = _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice
for 30 seconds
exten = _8X.,2, congestion() ; No answer, nothing
exten = _8X., 102, busy() ;
End result...
Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
to authenticate on INVITE to '6050
sip:[EMAIL PROTECTED];tag=as545ccba3'
SIP debug...
-- Executing Dial(SIP/6050-132b,
SIP/[EMAIL PROTECTED]|30) in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205
v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
(no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username=6050,realm=asterisk,nonce=42d82e9b,uri=sip:
[EMAIL PROTECTED],algorithm=MD5,response=420e39b35648a10c
129dd4fb5f97ec47
Contact: 6050 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED];tag=as2f065f18
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 64.4.192.110:5060
com*CLI
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
6 headers, 0 lines
com*CLI
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED];tag=SD38rq699-
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
WWW-Authenticate: DIGEST
RE: [Asterisk-Users] DTMF out to Cell Phone
Steve, The cellular system is Cingular, and as I said it works fine if the call is made from the cell phone to asterisk, so I don't think it's the cell switch, If the call is made through the asterisk box using a pri line, Digum T100P, to a cell phone then the DTMF does not work, for any application. Thanks for you response, John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Wednesday, March 09, 2005 7:55 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DTMF out to Cell Phone Hi John, You didn't say what kind of cellular system. If its an AMPS system (I don't think any other analogue cellular stiff exists) DTMF is quite troublesome. If it is a digital network the DTMF actually comes from the basestation, rather than the phone. Its is normally very high quality. However, its timing is nothing like the timing of the button pushes on the handset. The basestation stretches the digits to rather long ones. Possibly as much as a second each. Regards, Steve John Fullington wrote: I set up a monitoring system that calls my techs when a problem occurs on one of our networks, everything works fine unless asterisk calls a cell phone in which case the tech can not respond using dtmf. It works fine if the tech call in but not if asterisk call a tech's cell phone. Anyone one have any suggestions? Thanks John Fullington ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Hi John, You didn't say what kind of cellular system. If its an AMPS system (I don't think any other analogue cellular stiff exists) DTMF is quite troublesome. If it is a digital network the DTMF actually comes from the basestation, rather than the phone. Its is normally very high quality. However, its timing is nothing like the timing of the button pushes on the handset. The basestation stretches the digits to rather long ones. Possibly as much as a second each. [...] Content analysis details: (0.6 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO 0.5 URIBL_WS_SURBL Contains an URL listed in the WS SURBL blocklist [URIs: digium.com] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P slow getting line tone
On Wed, 2005-03-09 at 14:34 +0100, Fabrizio Mazzoni wrote: Hello all, I just installed a TDM400P with 2 FXO modules on my asterisk server. The card works perfectly. To get users to ring out from my SIP phones i setup an extension with 0 that basically does something like this: extension = 0,1,Dial(ZAP/g1) where g1 is the group of the two FXO channels extension = 0,2,Hangup This works exactly as i want so users basically can dial 0, wait for the dialtone and then dial the requested number. The only problem that i have is that from when a user dial 0 to when i get the dialtone from the telephone line, something like 5 seconds pass... is it possible to pull this wait time down to about 1 second? or even less?? Unless you are doing something odd that requires the user to listen to the dialtone and validate there is one, why don't you just go ahead and capture the number and dial it out. The benefit is that asterisk then logs the outgoing number and the times in CDR. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which hardware for this solution?
Hello, we are a firm who wants to develop some VOIP solutions. The first infrastucture we choose for development is: - an Asterisk machine connected to a traditional PBX (s0). In this way people is not (yet) obligated to migrate its extisting PBX (and analog phones) to VoIP. - The PBX will be then configured to redirect specific outgoing calls (i.e. a remote branch office) to Asterisk, that will deliver such call to a remote and previously configured Asterisk gateway (VPN). - The 'remote' Asterisk should converts this call, speak to the PBX connected and finally the phone ring. * Phone - Analog PBX - Asterisk - INTERNET - Asterisk - PBX - Phone * Straight to the point: what kind of hardware I need? I saw some PCI cards (like Digium Wildcard TE110P) but I am not sure what to buy. Any help, URLs is very much appreciated. Thanks in advance, Giorgio Mandolfo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Polycom IP600 Phantom Ringing
1.4.1 over here. Just to rule out all the possibilities - it's not MWI is it? http://www.voip-info.org/tiki-index.php? page=Getting%20MWI%20on%20Polycom%20Phones%20to%20work%20with%20Asterisk It shows up as a little half-ring. It should also be accompanied by the top LED flashing, and a stutter on the dialtone. Jerry wrote: Never had any of my 100 or so act like that. What version of code are you running? I think 1.4.1 is the latest. On Mar 8, 2005, at 8:05 PM, Ben Ruset wrote: Hello list: I have a very odd problem. Seemingly randomly, my Polycom IP600 phones will ring without a call being placed to it. That is to say, a random phone will ring. Nothing shows up under Caller ID. Even the buttons that light up to show an incoming call do not light up. If you pick up the handset, you can hear the phone ring through the speaker. Hanging up the phone makes it stop ringing. Then, sometime later, it will happen on another random extension. Is this a common problem? Where can I look to start diagnosing this? Thanks! -ben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk System() call error
On Wednesday 09 March 2005 13:37, Jonathan Hobbs wrote:
I have a linux (bash) script file which is invoked via:
exten = s,3,System(./BuildMsg.sh ${SFile} msg02 msg99)
When I start asterisk with the command: asterisk -gc
This script executes as expected ('asterisk -gc' and 'asterisk -vg' also
work). However, when I try to start asterisk with the command:
'asterisk -g' the script does not execute and I get the following error
message in the 'messages' log file:
Mar 9 08:06:55 WARNING[790]: Unable to execute './BuildMsg.sh 1-1 msg02
msg99'
Never put in relative paths. Try /etc/asterisk/BuildMsg.sh.
B
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Re: [Asterisk-Users] NuFone + VoIPJet = busy busy busy
On March 9, 2005 08:41 am, Jean-Michel Hiver wrote:
I'm using VoIPJet and NuFone as a fallback, and it seems that both of
them are circuit busy!
What exactly is the return code for nufone? Your dialplan should look
something like this:
exten = whatever,1,Dial([EMAIL PROTECTED]/${EXTEN},,g)
exten = whatever,2,NoOp(DIALSTATUS IS ${DIALSTATUS}, HANGUPCAUSE IS
${HANGUPCAUSE})
exten = whatever,3,Hangup
(obviously if you do other magic in your dialplan this needs to be adjusted.
The important part is the 'g' flag to Dial (go on after hangup), and the NoOp
which echos the dialstatus and hangupcause variables to the console.
Nufone is rock-solid stable. I have been using them for about 5kmin/month
over the past year with *no* issues, which is why I'd like to see what you're
getting back for a dialstatus and hangupcause.
-A.
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Re: [Asterisk-Users] Which box?
At the moment I'm just trying to figure out how the technology works - sort of proof of concept. Ideally, I'd like to move my business to such an environment at which time I'd definitely put the package on a dedicated box. Dunc Alistair Cunningham wrote: Dunc, Depends on the environment you run it in. If this is main telephony system for a business, then a dedicated machine is highly desirable, and you may also want to think about redundancy and failover. If it's for your own personal use, or it's a development machine, then it can co-exist with other software with no problems. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ asterisk wrote: I'm sure this is a stupid question, but I'm not finding an answer anywhere. Do I need a dedicated box to run asterisk, or can I put in my server (running Fedora) and leverage some of the free cpu cycles and disk space? Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail - No Audio Output!
The WAV files attached to the emails are recorded at a very low volume for whatever reason. Raise your volume higher and you'll hear them. -Herman Julius Kidubuka wrote: Hi all, I am able to receive voicemail in my mail box but when I try to play the audio file attachment, I hear nothing at all (yet the caller on the other end does leave a voicemail message)! Anyone had a similar problem before? Ideas are welcome! Note: I am using [EMAIL PROTECTED] 0.6 Thanks in advance, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk System() call error
On Wed, Mar 09, 2005 at 08:37:02AM -0500, Jonathan Hobbs wrote:
I have a linux (bash) script file which is invoked via:
exten = s,3,System(./BuildMsg.sh ${SFile} msg02 msg99)
Don't assume the daemon runs in a certain directory. The working
directory of a daemon should generally be '/' unless it is chrooted for
security concerns.
So please provide the full path to the script. Anyway, scripts don't
belong in /etc/asterisk. There is enough junk there already.
When I start asterisk with the command: asterisk -gc
This script executes as expected ('asterisk -gc' and 'asterisk -vg' also
work). However, when I try to start asterisk with the command:
'asterisk -g' the script does not execute and I get the following error
message in the 'messages' log file:
Mar 9 08:06:55 WARNING[790]: Unable to execute './BuildMsg.sh 1-1 msg02
msg99'
The script file is located in /etc/asterisk, and I have confirmed that
asterisk is looking for the script file in this location: I tried
exten = s,3,System(pwd location.out) and location.out contained
'/etc/asterisk'.
So let's start ruling out reasons:
1. use full path
2. System(ls -l /path/to/BuildMsg.sh /tmp/output)
3. System(strace -o /tmp/trace /path/to/BuildMsg.sh 1-1 msg02 msg99)
Asterisk is running as root and the group/owner of Buildmsg.sh are set to
'root' (I have also done a 'chmod 777 Buildmsg.sh' just to be sure). I am
running: Asterisk CVS-HEAD-02/17/05-11:17:10, on a linux box with GNU bash,
version 2.05b.0(1)
Distro? Kernel? Glibc?
--
Tzafrir Cohen | New signature for new address and | VIM is
http://tzafrir.org.il | new homepage | a Mutt's
[EMAIL PROTECTED] || best
ICQ# 16849755 | Space reserved for other protocols | friend
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Re: [Asterisk-Users] Voicetronix Tones
Hmmm, maybe the dtmfmode is incorrect. in your sip.conf what is dtmfmode set to? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call routing question
Hi Cameron,
Thanks for the suggestions. I think this is precisely what I was
looking for, unfortunately neither of those variables appears to be set
on my incoming calls. This is probably because I'm doing remote call
forwarding which is done by the phone company rather than regular call
forwarding. I guess I'll just have to get different numbers from my
VOIP provider in order to route my Verizon numbers to different extensions.
Thanks again for the help,
-Herman
Cameron Beattie wrote:
Try using the special identifiers ${DNID} or ${RDNIS}. Refer to http://www.voip-info.org/tiki-index.php?page=Asterisk%20variables for more info.
Regards
Cameron
Original message
--
Date: Tue, 08 Mar 2005 10:09:37 -0500
From: Herman Sheremetyev [EMAIL PROTECTED]
Subject: [Asterisk-Users] call routing question
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi All,
I have a question about call routing. I currently have a phone number provided by Voicepulse that connects directly to my Asterisk box and another phone number provided by Verizon that I have Remote Call
Forwarded to the Voicepulse number. What I'm wondering is if the
information about which number is actually dialed available for me to route the
calls to different extensions? Thanks for the help and I apologize if this has
already been discussed.
Thanks,
-Herman
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Re: [Asterisk-Users] Should ICMP port unreachable generate a BYE request?
You'd have to trace the code to work it out properly. But ICMP packets aren't generally passed to userspace. What's more likely is that the kernel, upon receiving sufficient of these errors, decides the connection is dead and notifies asterisk. That's what I'm thinking and just want to make sure. AFAIK, ICMP port unreach is generated when a UDP client tries to connect to a UDP server on a port with no server listening. In fact this technique is used by traceroute to accomplish its operation. So Asterisk will probably get some error value when it tries to write to a UDP socket and I wonder if it will generate a BYE as a result. It'll be great if I find it out without diving into the source and getting lost there :) Strange problem though, how can only some packets generate Port Unreachable, but not all. Random routing problem? Asterisk is running on a multihomed machine with one interface on providers network so there's no routing here, they're on the same LAN. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc error
In data Tue, 8 Mar 2005 18:25:38 + (GMT), hai scritto: [chan_zap.so]Mar 8 17:53:06 WARNING[2447]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_retrieve_call_to_death Mar 8 17:53:06 WARNING[2447]: loader.c:391 load_modules: Loading module chan_zap.so failed! I'm using asterisk from debian/sid: Asterisk 1.0.5-BRIstuffed-0.2.0-RC7e zaptel modules are version 1.0.4 zaphfc is from bristuff-0.2.0-RC7e and it's compiled against zaptel source version 1.0.4 - - - - - - 8 snipped Change your: load = chan_zap.so load = res_musiconhold.so ok, this worked. I had to load also chan_modem and i had to fix a missing [channels] in zapata.conf now i get: -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, Zap/g1/the number) in new stack -- Called g1/the number -- Channel 0/1, span 1 got hangup Mar 9 15:09:53 WARNING[5329]: app_dial.c:415 wait_for_answer: Unable to forward voice -- Hungup 'Zap/1-1' My asterisk setup works well with isdn4linux so i think that the problem relies in the zaphfc setup. ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to dial out using HFC ISDN card
I'm running * with a bog standard HFC ISDN card using zaphfc. Everything seems to work, including incoming calls, but I simply cannot make outgoing calls. This is very odd since the same card worked with the same configuration in another server. This is what I get from * debug. The only possible difference between the two servers that I can think of is that the HFC card is sharing an IRQ in the new server, whereas it wasn't in the old one. Could this possibly have an effect, and if so, how do I stop the card sharing IRQs with other devices? I've disabled everything that I don't need in the BIOS already. build_route: Contact hop: sip:[EMAIL PROTECTED] -- Executing Dial(SIP/extn702-82a5, Zap/1/01932567543) in new stack -- Called 1/01932567543 Ooh, format changed from unknown to alaw -- Channel 0/1, span 1 got hangup Unable to forward voice Set option AUDIO MODE, value: ON(1) on Zap/1-1 Hangup: channel: 1 index = 0, normal = 9, callwait = -1, thirdcall = -1 Not yet hungup... Calling hangup once with icause, and clearing call disabled echo cancellation on channel 1 Set option TDD MODE, value: OFF(0) on Zap/1-1 Updated conferencing on 1, with 0 conference users Set option AUDIO MODE, value: OFF(0) on Zap/1-1 disabled echo cancellation on channel 1 -- Hungup 'Zap/1-1' == No one is available to answer at this time Exiting with DIALSTATUS=NOANSWER. Thanks Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iaxy stopped working
Since yesterday the little iaxy does not register anymore! It is also not pingable with the last know IP address! Unplug plug in the power cable reacts in a short flashing of the Ethernet port and after a second or two a short flash of a green LED. What should I check next? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: how to sip-h323 using asterisk-oh323-0.7.1
i am using gnugatekeeper. i have three things gatekeeper ip, account, accountpassword how to set account and password in oh323.conf gatekeeper=gnu gatekeeper ip gatekeeperPassword=accountpassword accountCode=account is this ok any example how to use this i want to rout my sip call to this gatekeeper for h323. __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which box?
On Wed, Mar 09, 2005 at 12:58:05PM +, Alistair Cunningham wrote: Just out of interest, has anyone tried Asterisk @Home on User Mode Linux? IIRC it should not work, as user-mode-linux requires a special kernel. You can try to use its tarball . Anyway, I regularily test Rapid installation on QEmu . Performance indeed is far from native, but it works (as a reminder: qemu is from the guy who pulled the following cool stunt: http://fabrice.bellard.free.fr/tcc/tccboot.html ) -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 Phantom Ringing
On Tue, Mar 08, 2005 at 09:05:34PM -0500, Ben Ruset said: Hello list: I have a very odd problem. Seemingly randomly, my Polycom IP600 phones will ring without a call being placed to it. That is to say, a random phone will ring. Nothing shows up under Caller ID. Even the buttons that light up to show an incoming call do not light up. If you pick up the handset, you can hear the phone ring through the speaker. Hanging up the phone makes it stop ringing. Then, sometime later, it will happen on another random extension. Is this a common problem? Where can I look to start diagnosing this? You can run ethereal to capture all packets to / from the phone. Something is obviously causing this problem. If nothing shows up in ethereal, maybe there is a power problem. Are your phones POE or wall-wart? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Polycom IP600 Phantom Ringing
No. The phone itself physically rings as if it was getting a call. No lights light up, nor does it show a missed call, nor does it show CID. Noah Miller wrote: 1.4.1 over here. Just to rule out all the possibilities - it's not MWI is it? http://www.voip-info.org/tiki-index.php? page=Getting%20MWI%20on%20Polycom%20Phones%20to%20work%20with%20Asterisk It shows up as a little half-ring. It should also be accompanied by the top LED flashing, and a stutter on the dialtone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Thank you for the response. I still have the errors mentioned below, sip
response and Failed to authenticate on INVITE
[PP]
type=peer
username=PP
fromuser=PP
authuser=PP
fromdomain=sip.broadvoice.com
secret=XX
host=sip.broadvoice.com
dtmfmode=inband
insecure=very
context=sip
qualify=yes
disallow=all
allow=ulaw
allow=gsm
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=no
Does anyone else have any other suggestions?
On Wednesday 09 March 2005 06:56 am, MF Hulber wrote:
Try changing the extension from Broadvoice1 to the actual phone number
(and don't send your secret in a public email or maybe that's Chris'):
[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=XXX
username=8475100139
Zanzamar Majere wrote:
I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this response
(on call out). Any suggestions? I don't think it is a problem with the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my understanding.
Free world does work for calling out however. So I know at least that
works.
-- Got SIP response 400 Bad request back from 147.135.0.128
Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
to authenticate on INVITE to 'PP
sip:[EMAIL PROTECTED];tag=as5b80cade'
On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
First off... please cancel previous amplification request. I have
implemented your ideas with the same errored result.
I am not sure that we are not making it thru authentication. From my
digging and comparing packet dumps comparing the soft phone to asterisk
they have identical transactions through the ACK reply (the last one
on the debug below). The softphone seems to be authenticated after the
ACK. I am a newbie to debugging this stuff. I just want to get it
working.
Thanks everyone in advance for your help. I am certainly very very
happy to try anything.
Based on Luki's suggestions I...
Changed sip.conf...
[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=DELETED
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
Changed extensions.conf...
exten = _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice
for 30 seconds
exten = _8X.,2, congestion() ; No answer, nothing
exten = _8X., 102, busy() ;
End result...
Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
to authenticate on INVITE to '6050
sip:[EMAIL PROTECTED];tag=as545ccba3'
SIP debug...
-- Executing Dial(SIP/6050-132b,
SIP/[EMAIL PROTECTED]|30) in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205
v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
(no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username=6050,realm=asterisk,nonce=42d82e9b,uri=sip:
[EMAIL PROTECTED],algorithm=MD5,response=420e39b35648a10c
129dd4fb5f97ec47
Contact: 6050 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED];tag=as2f065f18
Call-ID: [EMAIL
Re: [Asterisk-Users] Polycom IP600 Phantom Ringing
They are all POE. Fed from a Cisco switch. Walt Reed wrote: You can run ethereal to capture all packets to / from the phone. Something is obviously causing this problem. If nothing shows up in ethereal, maybe there is a power problem. Are your phones POE or wall-wart? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: : RE: Re: MGCP to Inter Tel system
-Original Message- -this is very true, however, the current version of the Axxess software (9.0) supports SIP trunking natively on the IPRC. I just got my Axxess upgraded and am salivating to get * connected to it. Hmm, so 9.0 is out and it supports SIP natively. How did you plan to integrate the 2? -The Axxess will see the * as it would see an IP service provider. I don't know the specifics yet but it would probably be something like... -AXXESS(via IPRC port)* -The IPRC port in the Axxess is programmed up as a trunk and the dialplan just sends the traffic that the LCR programming determines is valid for that provider. Jason Kawakami www.optellabs.com Salt Lake City, UT ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX host=sip.broadvoice.com dtmfmode=inband insecure=very context=sip qualify=yes disallow=all allow=ulaw allow=gsm ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no Does anyone else have any other suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Help Site - cut down on Mailing List questions
Andrew Kohlsmith wrote: On March 9, 2005 03:49 am, Tzafrir Cohen wrote: The wiki has a section of exammple setups and configurations. What is the atvantage of your separate site? The wiki is very messy and hard to find information. And I say this as an experienced Asterisk user (multiple PRI setups, voicemail, spandsp, hard/softphones, manager interface, etc.) -- the wiki's a good idea but it is very... confusing? congested? I am all in favor of cutting down on Mailing list questions by having information available and organized. However, I'm not sure creating a new site helps in the long run. If the wiki is very messy and hard to find information, the focus should be on getting the wiki more organized and neat. I'm sure you could find some ways to help organize. Perhaps additional pages are needed to group things in a different way. Or maybe some pages could be changed to be better organized. Don Pobanz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone + VoIPJet = busy busy busy
Jean-Michel Hiver wrote: Hi List, I'm using VoIPJet and NuFone as a fallback, and it seems that both of them are circuit busy! How are you determining a fallback condition from one voip to another? greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Telecom echo cancel disable
Disabled echo canceller because of tone (tx) on channel 10 I understand that the PSTN companies use their own echo canceller's, send a tone across 2100hz, the problem we're having is people are complaining of echo on random calls. I'm assuming this may be the cause. Is their anyway to 'ignore' the disabling of EC? Or would be just be a manual code change.. Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice latest changes and still not working-An Additional Server
Jerry- Thank you I accidently sent my password on the LISTSERV last night so I just changed (pasted) the new one in. Still the same problem... Mar 9 09:51:13 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to 'Chris Nibeck sip:[EMAIL PROTECTED];tag=as4b70f2e7' Incoming works fine still. Anyone can call me at that number. Please do. It is a free call from another BV account. Chris On Mar 9, 2005, at 7:42 AM, Jerry Geis wrote: CHris, I had the exact same problem with the exact same error. My password was entered incorrectly in context section. The register line had the correct password. That is why you get incoming calls. and not outgoing. Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone + VoIPJet = busy busy busy
On March 9, 2005 10:43 am, Cirelle Internet Products wrote:
How are you determining a fallback condition from one voip to another?
Mine's rather simple but it works well:
[macro-nufone-dial]
exten = s,1,GotoIf($[$ACCOUNTCODE != ],s,gotac)
exten = s,n,SetVar(ACCOUNTCODE=${ARG2})
exten = s,n,GotoIf($[{$ARG2} != ],s,gotac)
exten = s,n,SetVar(ACCOUNTCODE=benshaw)
exten = s,n(gotac),SetAccount(${ACCOUNTCODE})
exten = s,n,GotoIf($[${LEN(${ARG1})} = 10]?s,add1)
exten = s,n,Dial(IAX2/[EMAIL PROTECTED]/${ARG1},,g)
exten = s,n,NoOp(NUFONE: HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
${DIALSTATUS})
exten = s,n,Goto(dial-${DIALSTATUS},1)
exten = s,n(add1),Dial(IAX2/[EMAIL PROTECTED]/1${ARG1},,g)
exten = s,n,NoOp(NUFONE: HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
${DIALSTATUS})
exten = s,n,Goto(dial-${DIALSTATUS},1)
exten = dial-CANCEL,1,Hangup
exten = dial-ANSWER,1,Hangup
exten = dial-NOANSWER,1,Hangup
exten = dial-BUSY,1,Busy
exten = dial-CONGESTION,1,Macro(pri-dial,${ARG1},${ARG2})
exten = dial-CHANUNAVAIL,1,Macro(pri-dial,${ARG1},${ARG2})
; handle NXX-NXX-, 1-NXX-NXX- and 011...
[nufone]
exten = _NXXNXX,1,Macro(nufone-dial,${EXTEN})
exten = _1NXXNXX,1,Macro(nufone-dial,${EXTEN:1})
exten = _011.,1,Macro(nufone-dial,${EXTEN})
You can ignore the accountcode stuff, we handle calls for several businesses
so I sort the accounting out that way.
For contexts that I want to have calls go out to Nufone I include the 'nufone'
context. As you can see, it handles 10-digit, 11-digit and international
(variable-digit) extensions.
Basically if it's a 10-digit #, add a '1' to it. Then attempt to Dial()
through my Nufone account. You'll notice the 'g' flag to the Dial()
application which tells it to go on in context after a hangup. I then check
the status of DIALSTATUS and if the result was CONGESTION or CHANUNAVAIL I
fall back and dial out my PRI.
Personally I think that CONGESTION should never be returned unless the other
side SAYS piss off, I'm too busy to handle your call but IAX will throw
back a CONGESTION status if it can't reach the other side, which is why I
have to check for both CONGESTION and CHANUNAVAIL.
-A.
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[Asterisk-Users] Cisco 7960 Protocol Invalid when Upgrading to 7.3
Cisco 7960 Upgrading from 6.x to 7.3 get Protocol Invalid. I'm sure this has been discussed but has anyone figured this out. Regards, Juan Staalenburg Teksavers, Inc. (512) 255-8395 x1002 AIM: juanteksavers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Thanks MF,
Yes that was me that sent my PW :-) It is changed now.
Same error...
Mar 9 10:12:46 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
to authenticate on INVITE to 'Chris Nibeck
sip:[EMAIL PROTECTED];tag=as0cefa74c'
Sip.conf...
[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=x
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
extensions.conf...
exten = _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice
for 30 seconds
exten = _8X.,2, congestion() ; No answer, nothing
exten = _8X., 102, busy() ;
On Mar 9, 2005, at 7:56 AM, MF Hulber wrote:
Try changing the extension from Broadvoice1 to the actual phone number
(and don't send your secret in a public email or maybe that's Chris'):
[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=XXX
username=8475100139
Zanzamar Majere wrote:
I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this
response
(on call out). Any suggestions? I don't think it is a problem with
the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my understanding.
Free world does work for calling out however. So I know at least that
works.
-- Got SIP response 400 Bad request back from 147.135.0.128
Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
to authenticate on INVITE to 'PP
sip:[EMAIL PROTECTED];tag=as5b80cade'
On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
First off... please cancel previous amplification request. I have
implemented your ideas with the same errored result.
I am not sure that we are not making it thru authentication. From
my digging and comparing packet dumps comparing the soft phone to
asterisk they have identical transactions through the ACK reply
(the last one on the debug below). The softphone seems to be
authenticated after the ACK. I am a newbie to debugging this
stuff. I just want to get it working.
Thanks everyone in advance for your help. I am certainly very very
happy to try anything.
Based on Luki's suggestions I...
Changed sip.conf...
[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=DELETED
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
Changed extensions.conf...
exten = _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial
Broadvoice for 30 seconds
exten = _8X.,2, congestion() ; No answer, nothing
exten = _8X., 102, busy() ;
End result...
Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response:
Failed to authenticate on INVITE to '6050
sip:[EMAIL PROTECTED];tag=as545ccba3'
SIP debug...
-- Executing Dial(SIP/6050-132b,
SIP/[EMAIL PROTECTED]|30) in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205
v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
(no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username=6050,realm=asterisk,nonce=42d82e9b,uri=sip:
[EMAIL PROTECTED],algorithm=MD5,response=420e39b35648a
10c 129dd4fb5f97ec47
Contact: 6050 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
Re: [Asterisk-Users] Cisco 7960 Protocol Invalid when Upgrading to 7.3
Cisco 7960 Upgrading from 6.x to 7.3 get Protocol Invalid. I'm sure this has been discussed but has anyone figured this out. See the Wiki. It's all there for ya. Don't recall the exact page name. Try searching on 7960 and brick. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Telecom echo cancel disable
Yeah. Edit zconfig.h and there's an option to ignore 2100hz. I didn't know what caused the 2100 until you said something. On Wed, 2005-03-09 at 09:47, Matt Schulte wrote: Disabled echo canceller because of tone (tx) on channel 10 I understand that the PSTN companies use their own echo canceller's, send a tone across 2100hz, the problem we're having is people are complaining of echo on random calls. I'm assuming this may be the cause. Is their anyway to 'ignore' the disabling of EC? Or would be just be a manual code change.. Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Help Site - cut down on Mailing List questions
On Wed, Mar 09, 2005 at 09:29:20AM -0600, Don Pobanz wrote: Andrew Kohlsmith wrote: On March 9, 2005 03:49 am, Tzafrir Cohen wrote: The wiki has a section of exammple setups and configurations. What is the atvantage of your separate site? The wiki is very messy and hard to find information. And I say this as an experienced Asterisk user (multiple PRI setups, voicemail, spandsp, hard/softphones, manager interface, etc.) -- the wiki's a good idea but it is very... confusing? congested? I am all in favor of cutting down on Mailing list questions by having information available and organized. However, I'm not sure creating a new site helps in the long run. If the wiki is very messy and hard to find information, the focus should be on getting the wiki more organized and neat. I'm sure you could find some ways to help organize. Perhaps additional pages are needed to group things in a different way. Or maybe some pages could be changed to be better organized. Or re-working the search mechanism. For instance: the ability to search in the titles of wiki pages alone. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
there are two of us with the same problem so I will answer for me. Yes I tried the below instructions. The current thinking by multiple people is * never tries authenticating so removing the FQDN will force * to go to the related section named by either a phone number or a non Fully Qualified Domain Name. But I still don't have it working so who knows. Anyone that wishes to call me via BV my number is 8475100139 and it is up. Chris On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote: Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX host=sip.broadvoice.com dtmfmode=inband insecure=very context=sip qualify=yes disallow=all allow=ulaw allow=gsm ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no Does anyone else have any other suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Print-to-Fax client
Hi, Does anyone know of a Print-to-Fax client that works with asterisk spandsp? Astfax is a partial solution but that only lets us email the fax in, we'ld like to set it up so the user can hit the print button and send the fax (even if all it does is email - transparently to the user - the fax to astfax). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.6 and chan_sccp problems?
Am I the only one seeing problems with chan_sccp and the latest Asterisk stable? Is there anyone where it is still working? My phones disappear after half an hour and are seen as dead by * Thx! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip hangup detection problem
Hi ml, I'm experiencing some problem detecting hangup with sip channel. I have an asterisk on remote site behind NAT and two xlite at home behind nat. I can make calls between them but hangup cannot be detected. When I try to hangup a call I see xlite that tell me hanging up for some seconds and hangups the call but the other side still be connected.. I also see on asterisk cli this message: chan_sip.c:787 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 45854 (Non-critical Response) Does someone experience the same problem? Can someone help me? Thanks. Marco Ziglioli ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF out to Cell Phone
Try setting your dtmfmode=inband in your your sip/iax/zap configs. This forces them to default to inband. You can then overide with info or whatever you need in other contexts. James On Wed, 09 Mar 2005 12:53:02 +0200, Mark Elkins [EMAIL PROTECTED] wrote: On Tue, 2005-03-08 at 14:16 -0500, John Fullington wrote: I set up a monitoring system that calls my techs when a problem occurs on one of our networks, everything works fine unless asterisk calls a cell phone in which case the tech can not respond using dtmf. It works fine if the tech call in but not if asterisk call a tech's cell phone. Anyone one have any suggestions? The application sounds interesting. Any chance you can email more about what you are actually doing? (code?) It sounds like your problem has nothing to do with mismatching Codec's or how the DTMF is being sent... etc... I have an Asterisk installation with BRI and with a premicell attached to an analogue interface (Premicell=fixed cell phone with analogue 2-wire interface that gives dial tone - like a trunk line) Perhaps I can then confirm your problem - or help with a solution? -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: how to sip-h323 using asterisk-oh323-0.7.1
I am using gnugatekeeper and asterisk.
My h323.conf:
[general]
bindaddr = ipaddress
tos=lowdelay
port = port
accountcode=xyz
gatekeeper = gatekeeper ipaddress
[xyz]
type=h323
prefix=123
context=default
extension.conf:
[default]
exten = _321,1,Dial(H323/${EXTEN:[EMAIL PROTECTED] ipaddress:port|30|r)
exten = _123,1,Dial(SIP/${EXTEN:3},30)
exten = _123,2,Voicemail(u${EXTEN:3})
exten = _123,102,Voicemail(u${EXTEN:3})
If I call 321, then asterisk route the call to the gatekeeper and gnugk
ring the extension.
gatekeeper.ini:
[Endpoint]
Gatekeeper=ipaddress of asterisk
Type=Gateway
H323ID=xyz
Prefix=123
[RasSrv::PermanentEndpoints]
ipaddress of asterisk=xyz;123
If I call 123, then gatekeeper route the call to asterisk, and asterisk
ring the extension.
--
Török József
Pharma-Chip Kft
1148, Budapest
Xantus u. 3.
Tel.: (1)-221-54-29
Fax: (1)-220-9415
http://www.pharmachip.hu
2005. március 9. 15.44 dátummal Kamran Ahmad ezt írta:
i am using gnugatekeeper. i have three things
gatekeeper ip, account, accountpassword how to set
account and password in oh323.conf
gatekeeper=gnu gatekeeper ip
gatekeeperPassword=accountpassword
accountCode=account
is this ok any example how to use this i want to rout
my sip call to this gatekeeper for h323.
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RE: [Asterisk-Users] Print-to-Fax client
I've seen a fax-printer driver for Windows PCs in the source (TurboPower's AsynchPro). Would be an interesting project to adapt it for * use. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 09, 2005 10:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Print-to-Fax client Hi, Does anyone know of a Print-to-Fax client that works with asterisk spandsp? Astfax is a partial solution but that only lets us email the fax in, we'ld like to set it up so the user can hit the print button and send the fax (even if all it does is email - transparently to the user - the fax to astfax). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 500 bitmaps and Idle Display Animation
Has anyone got this to work? Under Idle Display Animation, the administrators guide says "For example, a company logo could be displayed".. In the ipmid.cfg file, I enabled 'ind.idleDisplay.enabled' (ie changed it to 1), and under the IP 500 section, I added an entry for the bitmap that I want to display: bitmap.IP_500.66.name ="arf" but from there I'm not sure where to go...what do you change to tell the phone to actually use that bitmap on the main screen during idle conditions? Thanks Marty Mastera M3 Resources [EMAIL PROTECTED] Phone: 303.680.1283 x200 FAX: 206.666.1786 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice users...
It is my understanding from monitoring lists and reading press releases that on the BV BYOD plans, once you bump two paths (like a three-way call), the next path is at the per-minute rate. However, since I can only receive calls and can't seem to call out on BV, I can't test this... James Taylor On Tue, 8 Mar 2005 14:40:11 -0800, Dalon Westergreen [EMAIL PROTECTED] wrote: I do not believe that BV restricts the number of outgoing calls, but i did hear that there agreement states charge you for more then 4 simultaneous calls. I have also heard that they have not done this to date. --Dalon On Wed, 09 Mar 2005 08:18:50 +1100, Rod Bacon [EMAIL PROTECTED] wrote: Do broadvoice limit the number of concurrent calls that any given sip registrant can make? What about other similar providers? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Edit MGCP response
Hi there, I'd like to know if there's any way I can edit the fields asterisk sends in an MGCP response to my devices, without having to mess with the source code. What happens is that asterisk sends an F parameter in an audit endpoint message I don't want it to send. Does anyone know I can solve this? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GotoIf problem
Removing the quotes and eliminating s,3,gotoif did work but its not
what I am looking for.
What I want to do is the following: If a ani that comes in has 10
digits I want to change the ${CALLERIDNUM} to unknown. If the ani
is 10 digits just goto voicemail.
When I set up my [vmail] to look like below, it does not work. When I
send a 4 digit
ani my e-mail confirmation of the voicemail shows the 4 digit ani and
not Unknown.
[globals]
Setvar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM})} != $DIGITS}]?4:5)
exten = s,4,Setvar(${CALLERIDNUM}=Unknown)
exten = s,5,Voicemail(u${ext})
exten = s,6,Hangup
Kurt
On Wed, 09 Mar 2005 07:34:24 +1100, Howard Lowndes [EMAIL PROTECTED] wrote:
On Wed, 2005-03-09 at 05:29, kurt x wrote:
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding is
this venture. When I dial and access voicemail with an ani of 3000
the gotoif statement does not push the call to s|6. Its goes through
each line( 1,2,3,4,5,6,7) . In additon when I dial with a 10 digit
ani the s,3,Gotoif does not work. It also goes through each line(
1,2,3,4,5,6,7)
Any help is greatly appreciated.
Have you tried removing the quotes?
Thanks
Kurt
Asterisk CVS-HEAD-07/14/04-16:28:29 built by
[EMAIL PROTECTED] on a i686 running Linux
[globals]
${ext}=0
SetGlobalVar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM}}) = ${DIGITS}]?s|5)
exten = s,4,GotoIf($[${CALLERIDNUM} = 3000]?s|6)
exten = s,5,Voicemail(u${ext})
exten = s,6,Background(pbx-invalid)
exten = s,7,Hangup
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Fwd: Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server ****SOLVED****
This configuration solved my problem. I could have sworn I tried this before. I guess not. I did not need to apply the patch. Also, I am using a regular Registration setup in my sip.conf not broadvoice's funky one... The only thing I can surmise is that order of the variables matters. This is what worked for me: [PP] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=PP secret=XX username=PP insecure=very context=sip authname=PP dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no Thank you On Wednesday 09 March 2005 08:23 am, Mike Matthews wrote: Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX host=sip.broadvoice.com dtmfmode=inband insecure=very context=sip qualify=yes disallow=all allow=ulaw allow=gsm ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no Does anyone else have any other suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which hardware for this solution?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Mandolfo Sent: Wednesday, March 09, 2005 8:59 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which hardware for this solution? Hello, we are a firm who wants to develop some VOIP solutions. [...] Straight to the point: what kind of hardware I need? I saw some PCI cards (like Digium Wildcard TE110P) but I am not sure what to buy. You need to but the appropriate cards to interface with the PBX you are trying to connect to. Without knowing what interfaces it has available, that's a difficult question to answer. If it's got an E1 or T1 interface, buy an appropriate port-density T1/E1 card (surprise) like a TE110P or TE410/405P. If it's analog, and appropritaely-configured TDM400P would be the way to go. Cards are cardsget what you need to make the interface happen. It's like asking what card you need to connect your computer to some undescribed network. If the network is ethernet, you need an ethernet card. If it's token ring, you need a token ring card, etc. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GotoIf problem
kurt x wrote: [globals] Setvar(DIGITS=10) Try this instead... [globals] DIGITS=10 -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk System() call error *SOLVED*
The full path name fixed the problem with script execution.
Thanks for all the help!
Jonathan
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: March 9, 2005 9:16 AM
Subject: Re: [Asterisk-Users] Asterisk System() call error
On Wed, Mar 09, 2005 at 08:37:02AM -0500, Jonathan Hobbs wrote:
I have a linux (bash) script file which is invoked via:
exten = s,3,System(./BuildMsg.sh ${SFile} msg02 msg99)
Don't assume the daemon runs in a certain directory. The working
directory of a daemon should generally be '/' unless it is chrooted for
security concerns.
So please provide the full path to the script. Anyway, scripts don't
belong in /etc/asterisk. There is enough junk there already.
When I start asterisk with the command: asterisk -gc
This script executes as expected ('asterisk -gc' and 'asterisk -vg' also
work). However, when I try to start asterisk with the command:
'asterisk -g' the script does not execute and I get the following
error
message in the 'messages' log file:
Mar 9 08:06:55 WARNING[790]: Unable to execute './BuildMsg.sh 1-1 msg02
msg99'
The script file is located in /etc/asterisk, and I have confirmed that
asterisk is looking for the script file in this location: I tried
exten = s,3,System(pwd location.out) and location.out contained
'/etc/asterisk'.
So let's start ruling out reasons:
1. use full path
2. System(ls -l /path/to/BuildMsg.sh /tmp/output)
3. System(strace -o /tmp/trace /path/to/BuildMsg.sh 1-1 msg02 msg99)
Asterisk is running as root and the group/owner of Buildmsg.sh are set
to
'root' (I have also done a 'chmod 777 Buildmsg.sh' just to be sure). I
am
running: Asterisk CVS-HEAD-02/17/05-11:17:10, on a linux box with GNU
bash,
version 2.05b.0(1)
Distro? Kernel? Glibc?
--
Tzafrir Cohen | New signature for new address and | VIM is
http://tzafrir.org.il | new homepage | a Mutt's
[EMAIL PROTECTED] || best
ICQ# 16849755 | Space reserved for other protocols | friend
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Re: [Asterisk-Users] DID in the U.S.
TEXAN services are provided by special tariff for State agencies in Texas. This is not available to the general public. The numbers are FREE because the State spends $$$,$$$,$$$,$$$.00. And this is only for the PLEXAR (enhanced centrex) type service. There are Free DID numbers with some vendors - you pay .01 min. There are Free DID numbers - you pay for bandwidth. There are Free DID numbers - you pay for rack space. There are $1 DID numbers - .008 min There are $19.95 DID numbers (Broadvoice, Packet8, and others) with no per minute fee. What you pay for numbers is not the question here. It's how much the whole thing costs: numbers, transport, switching, access, entrance facility, bandwidth, rackspace... There are lots of rate elements that are factored into the cost of service. Most of the lower cost or Free numbers requires a long (30-60 days) lead-time to setup, a long term commitment, and enough monthly volume to make someone happy. James Taylor On Tue, 08 Mar 2005 16:47:19 -0600, Doug Millsaps [EMAIL PROTECTED] wrote: At 04:15 PM 3/8/2005, you wrote: Hello! Have a look at the following page: http://www.tex-an-2000.com/plxr.html Block of 10.000 DID numbers: No charge Is there something comparable in the LA area? Andreas I believe it's only free if you pay for the other services listed on that page. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GotoIf problem
To me it looks like the $LEN function is not working. When I do verbose start to * I see that it walks right through every step whether or not the ani is 10 digits or something else. Would it be better to write an AGI script? Kurt On Wed, 09 Mar 2005 11:41:50 -0600, Chris Wade [EMAIL PROTECTED] wrote: kurt x wrote: [globals] Setvar(DIGITS=10) Try this instead... [globals] DIGITS=10 -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildcard X100P or TDM400P?
The only reason i like this card, it is a cheap RTC which we need for conferencing. We also used ztdummy but Dell Poweredge servers we use, have ohci USB and ztdummy only works with uhci. I wish some one would come up with some other means to have RTC. :-( On Tue, 2005-03-08 at 12:30 -0600, Steven Critchfield wrote: On Tue, 2005-03-08 at 09:56 -0800, Spencer Nassar wrote: I'm looking to add a single FXO port to my Asterisk box. It looks like my options are a Digium Wildcard X100P off eBay for $6.99, or a Wildcard TDM400P with an FXO Module from Digium for $125. Can anyone explain the tradeoffs (other than the ability to put 4 FXO/FSO modules on the TDM400P). What about RTC for the system - I know the TDM400P provides it. Does the X100P? The X100P you are seeing on Ebay are not from Digium. They do not come with support time from Digium. They seem to only support US style analog lines(600ohm). If you have trouble, you are most likely on your own. TDM400P with FXO daughter card includes 1 hour of Digium support. It is supposed to support other line types. If you have trouble, it is likely you will get direct support from Digium and from the community here. TDM400P card is also capable of adding more ports without increasing interupts on your server. If you need to add another X100P later on, that will double the interupt load on the machine. With no one really making much money on the X100P card and I don't think anyone is making them anymore, you may not get new features added to the driver. The TDM400P card will probably be developed for a while to come as it the current option. -- Alex Litvak [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] max number of conference rooms, and max number of conference callers in one room
Hi Guys, Does anyone have knowledge about max number of conference rooms, and max number of conference callers in one room? Thank you so much. jintwo __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GotoIf problem
kurt x wrote: To me it looks like the $LEN function is not working. When I do verbose start to * I see that it walks right through every step whether or not the ani is 10 digits or something else. Would it be better to write an AGI script? Kurt I use LEN quite a lot, works perfectly. Go back NoOp EVERYTHING, every single value, variable, etc... Run the logic through your head, if your head says it should work but it doesn't, we'll need to look into it again. Otherwise, NoOp'ing everything will help you and the rest of us debug what is happening. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Print-to-Fax client
On Wed, 2005-03-09 at 11:31 -0500, [EMAIL PROTECTED] wrote: Hi, Does anyone know of a Print-to-Fax client that works with asterisk spandsp? Astfax is a partial solution but that only lets us email the fax in, we'ld like to set it up so the user can hit the print button and send the fax (even if all it does is email - transparently to the user - the fax to astfax). Turn off HTML. Research cups. I know people who use cups to create PDFs for them. Shouldn't be any really big effort to get cups and windows to talk to each other and create a print job that creates the .call file and the ,ps file as well. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GotoIf problem
kurt x wrote:
[globals]
Setvar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM})} != $DIGITS}]?4:5)
exten = s,4,Setvar(${CALLERIDNUM}=Unknown)
exten = s,5,Voicemail(u${ext})
exten = s,6,Hangup
Oh, and it should be
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM})} != ${DIGITS}]?4:5)
-Chris
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[Asterisk-Users] Assistance with Overhead Paging
We have purchased an Asterisk based PBX solution that is completely setup with one exception, overhead paging. We have a powered Paging System and want to find a way to set an extension (i.e. 999) to use the sound card, which would in turn go out the paging system speakers. I've seen several references to this being possible, but no examples how to do it. The system we purchased does not provide the sound card as an option for an extension, only SIP phone, analog phone, IVR, Call Queue, Voice Mail, Agent log-in, Agent log-out Any help would be appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
