[asterisk-users] Remote extension search?
I've heard about this, but I really can't seem to find anything on it. I've got a strange setup that exists only because of firewall issues, and everything about it seems fine. The setup: SIP clients - Asterisk (office) - IAX - Asterisk (colocation) - SIP PSTN Termination All the extensions I want to be able to dial are on the colocation box. What I'd really like is for the office asterisk box to forward all extension requests it doesn't know about to the colocation Asterisk box. I think this is refered to as Trunking. I only need to do this in a single direction, if that's any easier to setup. Are there any good documents on VOIP-Info or another site on setting up something like this? The office Asterisk's job is just to act as a SIP to IAX gateway. I've got a work-a-round that will work, but I thought I'd learn the proper method. -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote extension search?
I used the macro-stdextenion that comes with every Asterisk install, and added a new option - s-CHANUNAVAIL which then dialled the other server via IAX. Worked really well and only took a few minutes. PaulH On Tue, 2007-08-14 at 23:51 -0700, Nicholas Blasgen wrote: I've heard about this, but I really can't seem to find anything on it. I've got a strange setup that exists only because of firewall issues, and everything about it seems fine. The setup: SIP clients - Asterisk (office) - IAX - Asterisk (colocation) - SIP PSTN Termination All the extensions I want to be able to dial are on the colocation box. What I'd really like is for the office asterisk box to forward all extension requests it doesn't know about to the colocation Asterisk box. I think this is refered to as Trunking. I only need to do this in a single direction, if that's any easier to setup. Are there any good documents on VOIP-Info or another site on setting up something like this? The office Asterisk's job is just to act as a SIP to IAX gateway. I've got a work-a-round that will work, but I thought I'd learn the proper method. -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi limitation?
Greetings list,
I've been using DUNDi for some time now to prevent calls between users going
out via PSTN if there's no need, set up as follows:
[macro-dundi-e164]
exten = s,1,Goto(${ARG1},1)
include = dundi-e164
[dundi-e164]
include = in-e164
switch = DUNDi/e164
My outbound call macro tries [macro-dundi-e164] first, if that fails then goes
onto PSTN connectivity.
However, I noticed a strange error last night when programming up a simple IVR
for a new customer. I added the following to [in-e164]:
exten = number,n,Background(ivr/hello)
exten = number,n,WaitExten(5)
; service
exten = 1,1,Macro(queue, service,300)
; accounts
exten = 2,1,Macro(queue,accounts,300)
exten = t,1,Goto(number,1)
(e164 number removed to protect client's privacy)
This works fine when calling externally, but when calling internally via DUNDi,
it's impossible to hit either IVR button because the call appears to still be
in the originating context rather than [in-e164].
I tried getting round this by changing the first line in [macro-dundi-e164]
from:
exten = s,1,Goto(${ARG1},1)
to:
exten = s,1,Goto(in-e164,${ARG1},1)
However, this breaks all external calling with an invalid extension error.
Can anyone suggest a way round this?
Thanks in advance.
Regards,
Chris
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Re: [asterisk-users] Remote extension search?
On Tue, 14 Aug 2007, Nicholas Blasgen wrote:
I've heard about this, but I really can't seem to find anything on it. I've
got a strange setup that exists only because of firewall issues, and
everything about it seems fine. The setup:
SIP clients - Asterisk (office) - IAX - Asterisk (colocation) - SIP PSTN
Termination
All the extensions I want to be able to dial are on the colocation box.
What I'd really like is for the office asterisk box to forward all
extension requests it doesn't know about to the colocation Asterisk box. I
think this is refered to as Trunking. I only need to do this in a single
direction, if that's any easier to setup.
You can do this by having a default widlcard extension to point to the
co-lo box. Simple (crude?) but effective... So in the diaplan in the
office asterisk box:
exten = _X.,1,Dial(IAX2/co-lo/${EXTEN})
so anything not matched locally will get punted through to the co-lo box.
This assumes 'co-lo' has an entry in iax.conf. (and corresponding
authentication on the co-lo side)
You can narrow it down if you know the extension numbering scheme - so if
you want to point all numbers starting with 0 and all 3-digit extensions
in the range 200 through 399, to the co-lo, then:
exten = _0.,1,Dial(IAX2/co-lo/${EXTEN})
exten = _2XX,1,Dial(IAX2/co-lo/${EXTEN})
exten = _3XX,1,Dial(IAX2/co-lo/${EXTEN})
and so on.
Although the last 2 might be combined with:
exten = _[23]XX,1,Dial(IAX2/co-lo/${EXTEN})
etc.
If you explicitly wanted to dial 9 for an outside line, then:
exten = _9.,1,Dial(IAX2/co-lo/${EXTEN:1})
Are there any good documents on VOIP-Info or another site on setting up
something like this? The office Asterisk's job is just to act as a SIP to
IAX gateway. I've got a work-a-round that will work, but I thought I'd
learn the proper method.
With so many ways to do something, who defines proper :)
Gordon
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Re: [asterisk-users] Patent issues, what features we can't use?
Zeeshan, First off, if your fear of being sued is what stops you from doing business then get out of the industry or get over it. Its a risk we all take everyday (not just in VoIP). You build up a core of Insurance and Defensive Patents to protect yourself. Risk is just part of doing business. Elements of the Asterisk that are clearly incompatible with the Dual License model are not included in the regular distribution. You may find them as add-on modules or in Trunk (If it supports a free development/education license) but not as a part of the regular distribution. To address the real issue... In the USA in recent years companies have been granted broadly worded patents. People at the patent office are clerks and not engineers. Plus, they have to deal with ALL INDUSTRY (e.g. Medical, Aviation, Computer Science, Earth Science, Early Childhood Development, Mining, Agriculture, Automotive, Maritime, Textile, Nuclear Physics, Beauticare, Electronics, Chemistry, Mechanics, Pharmaceutical, etc...etc...etc...) not just Telecom. It is quite literally impossible to understand enough about everything to make clear judgments as to what is truly patentable and what is not. The patent office position is basically Spell it out to us and let the courts figure out the rest. While most broadly worded patents are unenforceable it still takes a legal process to get the patents dismissed as too vague. That process can be VERY costly for the person sued as well as the suer (sp). For a large telecom is all just part of the cost of doing business. Most smaller companies (e.g. us guys) are forced to settle because we haven't the millions of dollars needed to defend ourselves. Now that being said where does the g729 patent (and the like) fit in? A patent like g729 is actually VERY specific about what it does and how to do it. Sure its a software patent but there is little room in the wording about what it accomplishes, by what means and the limitations of the patent. Plus the price is very reasonable at $10/channel (non-transcoding pass through requires no License). Additionally, g729 is not the only game in town when it comes to low-bandwidth codecs. (Personally I like to use g726-32 its lightweight and transcodes to/from uLaw easily...but I digress). This varies from some other software patents for One-Click-Checkout or Online Shopping Cart. They are both patented and every challenge has been settled out of court, thus they still stand a viable patents. Ultimately the question comes down to...Do you want to stay home and hide or would you rather come out and play? Just my input, Mark C. Zeeshan Zakaria wrote: Hi everybody, As the Asterisk community is getting larger and larger, I was wondering that the features which are provided in Asterisk and are programmed by the open source community under GPL, or GUIs like FreePBX which also come loaded with wonderful features and uses same Asterisk, are they anywhere violating any patent laws? Most of the features work the same way as Nortel, Avaya and other PBX systems. Is there anyone who owns these features and will come one day to claim his royalties? When I deploy an asterisk soultion for a customer, is there any violation of any patent or copyright laws anywhere? Of if I use my own Asterisk server to provide services to some customers, am I violating any patent laws by not paying the royalties to some patent owners? I heard people saying that IVR technology is patented and google search for patents also say so. But we all are using IVR for ourselves and our customers without paying royalties to anyone. But when it comes to using g729, all of a sudden royalty issue comes in. So what is right to use and what is not? -- Zeeshan A Zakaria -- As I slowly sip my coffee I feel my humanity start to slip back into me and realize what a foul beast humanity really is. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Digium TE120P card support MFCR2
On Mon, 2007-08-13 at 16:15 +0530, [EMAIL PROTECTED] wrote: Hi, I have successfully configured DIGIUM card and successfully communicated through it to the another E1 card running application. Can anybody tell me does TE120P support MFC/R2 protocol. As far as I know the card is not the issue. But you need to install additional software. Basically you need spandsp, libunicall, libmfcr2, libsupertone and chan_unicall from http://www.soft-switch.org If you use RHEL or CentOS then you can find a set of Asterisk 1.2 SRPMs that support this already at: http://www.laimbock.com/dl/asterisk/ You will need to rebuild the SRPMs with the options enabled that you require. Have a look in zaptel.spec and asterisk.spec. Regards, Patrick (remove -list from my email address if you want to reply privately) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Can you reload only one conf file?
Hi Mike, Consider ARA www.voip-info.org/wiki/index.php?page=Asterisk+RealTime www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions or you can use dialplan add extension cli command from Asterisk Manager Interface. see http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action +Command Regards Nasir Iqbal ICT Innovations On Fri, 2007-08-10 at 12:19 -0400, Mike wrote: Well, if you really must know (this is OT for everybody else I guess) I have a custom Web GUI used for my customers, and when some settings are modified, a conf file is created. This conf file must be reloaded at this point, therefore I call the reload command externally. Why do I do this? Because the %*$%/$ hint fonctionnality can't accommodate variables fetched from a DB like the rest of my dialplan. Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] why is nonce=584760da used in sip packets?
Hi all, There is a parameter called nonce included in every register request that a UA sends to asterisk. I have read sip debug a lot and only found out that the nonce parameter value which is used in register request was generated by asterisk server in a previous sip response. As you can see in the sip debug (labled in red). --- Transmitting (NAT) to 208.120.167.146:80 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 208.120.167.146:80;branch=z9hG4bK03891485;received= 208.120.167.146 From: sip:[EMAIL PROTECTED];tag=as65460c44 To: sip:[EMAIL PROTECTED];tag=as3a5cc850 Call-ID: [EMAIL PROTECTED] CSeq: 19680 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=584760da Content-Length: 0 Scheduling destruction of SIP dialog ' [EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) magnum*CLI --- SIP read from 208.120.167.146:80 --- REGISTER sip:magnum.axvoice.com SIP/2.0 Via: SIP/2.0/UDP 208.120.167.146:80;branch=z9hG4bK0c6c6f53 From: sip:[EMAIL PROTECTED];tag=as65460c44 To: sip:[EMAIL PROTECTED];tag=as3a5cc850 Call-ID: [EMAIL PROTECTED] CSeq: 19681 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=bernart48, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED]:9060, nonce=584760da, response=948d3923bf2df47eca17c572713af2c7, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED]:80 Event: registration Content-Length: 0 What i dont know, and would very much like to know, is what is the purpose of this parameter in sip packets? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why is nonce=584760da used in sip packets?
- Rizwan Hisham [EMAIL PROTECTED] wrote:
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=584760da
Authorization: Digest username=bernart48, realm=asterisk, algorithm=MD5,
uri=sip:[EMAIL PROTECTED]:9060, nonce=584760da,
response=948d3923bf2df47eca17c572713af2c7, opaque=
What i dont know, and would very much like to know, is what is the
purpose of this parameter in sip packets?
It's kind of challenge algorithm. What you see in response is not
MD5(password), but MD5('password', 'realm', ..., 'nonce'). Nonce is generated
by server so that you don't get the same hash for for every authorization by
that user. It prevents someone who can see only one way communication from
breaking your sip session + makes breaking hash a little bit harder.
Nonce should be unique per authorization.
If nonce wasn't used you could reuse the same response in next connection even
if you don't know the real password.
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[asterisk-users] Dundi x ENUM
Hi all, I've just being wondering if Dundi has the same purpose as ENUM. I don't know much (actually almost nothing) about these technologies. As far as I know they are a kind of DNS resolver used in the VoIP context. For example, user [EMAIL PROTECTED] has the extension namber 1001. This way nobody has to know the ronaldo's extension number. I'll appreciate if someone can clear my understanding about that? Thanks in advance. Ronaldo. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why is nonce=584760da used in sip packets?
thanx for the reply. what i have understood from ur reply and from googling
is that for every authorisation there is a unique nonce (or new nonce), and
previous nonce is expired. but i have seen in sip debug on my atserisk cli
that :
for the first register request, server sends an unauthorisation response
with a new nonce like below:
REGISTER sip:magnum.axvoice.com SIP/2.0
Via: SIP/2.0/UDP 208.120.167.146:80;branch=z9hG4bK722c974c
From: sip:[EMAIL PROTECTED];tag=as1acc7245
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 19710 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username=bernart48, realm=asterisk, algorithm=MD5,
uri=sip:magnum.axvoice.com, nonce=325611ed,
response=8105b402d3b955cb65bd9aa8e498cbc8, opaque=
Expires: 120
Contact: sip:[EMAIL PROTECTED]:80
Event: registration
Content-Length: 0
--- Transmitting (NAT) to 208.120.167.146:80 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 208.120.167.146:80;branch=z9hG4bK722c974c;received=
208.120.167.146
From: sip:[EMAIL PROTECTED];tag=as1acc7245
To: sip:[EMAIL PROTECTED];tag=as1d329593
Call-ID: [EMAIL PROTECTED]
CSeq: 19710 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1312f4b5
Content-Length: 0
Now in the second register request, the nonce which should be used is the
one found in latest unauthorisation response recieved from server, but in
this case the nonce used is from previous unauthorisation response as shown
below.
--- SIP read from 208.120.167.146:80 ---
REGISTER sip:magnum.axvoice.com SIP/2.0
Via: SIP/2.0/UDP 208.120.167.146:80;branch=z9hG4bK722c974c
From: sip:[EMAIL PROTECTED];tag=as1acc7245
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 19710 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username=bernart48, realm=asterisk, algorithm=MD5,
uri=sip:magnum.axvoice.com, nonce=325611ed,
response=8105b402d3b955cb65bd9aa8e498cbc8, opaque=
Expires: 120
Contact: sip:[EMAIL PROTECTED]:80
Event: registration
Content-Length: 0
This causes the asterisk server to send another unauthorisation response
with an additional parameter stale in WWW-Authenticate section as shown
below
--- Transmitting (NAT) to 208.120.167.146:80 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 208.120.167.146:80;branch=z9hG4bK722c974c;received=
208.120.167.146
From: sip:[EMAIL PROTECTED];tag=as1acc7245
To: sip:[EMAIL PROTECTED];tag=as1d329593
Call-ID: [EMAIL PROTECTED]
CSeq: 19710 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=4f90fab4,
stale=true
Content-Length: 0
this stale=true field causes the asterisk server to display the following
NOTICE on the cli
NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale
nonce received from 'sip:[EMAIL PROTECTED]'
and this will continue happening unless the next register request uses the
nonce field recieved in latest unauthorisation response from server, and
untill then the user agent will not be able to register with the server.
This will cause problems in our services.
I hope u understand the problem. Sorry for this very long reply. If you know
how to deal with this problem then plz share ur solution. I have been facing
this problem for 2 weeks now and uptill now i only have found out the reason
for this problem. Now is the time for search for the solution.
Hope to hear from u soon
On 8/15/07, Stanisław Pitucha [EMAIL PROTECTED] wrote:
- Rizwan Hisham [EMAIL PROTECTED] wrote:
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=584760da
Authorization: Digest username=bernart48, realm=asterisk,
algorithm=MD5, uri=sip:[EMAIL PROTECTED]:9060, nonce=584760da,
response=948d3923bf2df47eca17c572713af2c7, opaque=
What i dont know, and would very much like to know, is what is the
purpose of this parameter in sip packets?
It's kind of challenge algorithm. What you see in response is not
MD5(password), but MD5('password', 'realm', ..., 'nonce'). Nonce is
generated by server so that you don't get the same hash for for every
authorization by that user. It prevents someone who can see only one way
communication from breaking your sip session + makes breaking hash a little
bit harder.
Nonce should be unique per authorization.
If nonce wasn't used you could reuse the same response in next connection
even if you don't know the real password.
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Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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Re: [asterisk-users] Asterisk DTMF Tones
Same problem. I've tested a Linksys/PAP2-3.1.9(LSc) and tried with INBAND configuration in both, asterisk and linksys EP, and it works. But, just was a test, dont know if I would let it in INBAND config. Lastly I tried with INFO in linksys, and rfc2833 in Asterisk, and works too.., no problem. On 8/3/07, Keshav K. [EMAIL PROTECTED] wrote: I have used Asterisk 1.2 and 1.4 with ATAs and PAP2. There is no issue in that. For that confrim to your service provider that whihc they accepts, invand or rfc Keshav John Meksavan [EMAIL PROTECTED] wrote: Asterisk Users, I am running Asterisk 1.2.13 on Debian Linux 2.6.18-4-amd64 and having problems with DTMF Tones. I have sip service from Teliax and configure to use rfc2833 for dtmfmode. The problem occurs, when I am using Linksys PAP2T phone adapter with a regular analog phone. Is this an issue with Asterisk? Or the Linksys PAP2Any insights would be greatly appreciated. Best Regards, John _ http://liveearth.msn.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Kesh Lets change the future...lets change the world. Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- caio ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dundi x ENUM
On 08:33, Wed 15 Aug 07, Ronaldo wrote: Hi all, I've just being wondering if Dundi has the same purpose as ENUM. I don't know much (actually almost nothing) about these technologies. As far as I know they are a kind of DNS resolver used in the VoIP context. For example, user [EMAIL PROTECTED] has the extension namber 1001. This way nobody has to know the ronaldo's extension number. I'll appreciate if someone can clear my understanding about that? The two are totally different. Dundi is used to provide callrouting information between asterisk boxen. For more information: http://en.wikipedia.org/wiki/DUNDI Enum is a way to not having to remember all my phone numbers, email addresses, IM accounts etc. Basically you get e phonenumber, and a special DNS record maps that phonenumber to more contact information. More information: http://en.wikipedia.org/wiki/ENUM -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why is nonce=584760da used in sip packets?
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham Sent: Wednesday, August 15, 2007 7:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] why is nonce=584760da used in sip packets? This causes the asterisk server to send another unauthorisation response with an additional parameter stale in WWW-Authenticate section as shown below --- Transmitting (NAT) to 208.120.167.146:80 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 208.120.167.146:80;branch=z9hG4bK722c974c;received=208.120.167.146 From: sip:[EMAIL PROTECTED];tag=as1acc7245 To: sip:[EMAIL PROTECTED];tag=as1d329593 Call-ID: [EMAIL PROTECTED] CSeq: 19710 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=4f90fab4, stale=true Content-Length: 0 this stale=true field causes the asterisk server to display the following NOTICE on the cli NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce received from 'sip:[EMAIL PROTECTED] ' and this will continue happening unless the next register request uses the nonce field recieved in latest unauthorisation response from server, and untill then the user agent will not be able to register with the server. This will cause problems in our services. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why is nonce=584760da used in sip packets?
You have on your hands a broken UA, since it is not responding to the changing nonce value. - Brad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham Sent: Wednesday, August 15, 2007 7:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] why is nonce=584760da used in sip packets? --- Transmitting (NAT) to 208.120.167.146:80 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 208.120.167.146:80;branch=z9hG4bK722c974c;received=208.120.167.146 From: sip:[EMAIL PROTECTED];tag=as1acc7245 To: sip:[EMAIL PROTECTED];tag=as1d329593 Call-ID: [EMAIL PROTECTED] CSeq: 19710 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=4f90fab4, stale=true Content-Length: 0 this stale=true field causes the asterisk server to display the following NOTICE on the cli NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce received from 'sip:[EMAIL PROTECTED] mailto:sip:[EMAIL PROTECTED] ' and this will continue happening unless the next register request uses the nonce field recieved in latest unauthorisation response from server, and untill then the user agent will not be able to register with the server. This will cause problems in our services. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma Wanpipe installation problems
I'm trying to install wanpipe on my new 2.6.21-2-amd64 core 2 duo machine (Debian's amd64 works with EMT64 too) to run Asterisk. I'm getting compilation errors when trying to install the wanpipe utilities. We are intending to use a Sangoma A102 card for ISDN30 in the UK. I've tried both the current stable 2.3.4-10 release and the new wanpipe-3.1.3 beta release of wanpipe. The first fails due to the recent kernel version. The second fails as follows: /usr/bin/ld: i386 architecture of input file `../lib/hdlc/wanpipe_hdlc.o' is incompatible with i386:x86-64 output Any help gratefully received. Rory -- Rory Campbell-Lange Director Campbell-Lange Workshop Ltd. [EMAIL PROTECTED] www.campbell-lange.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disable MoH for certain phones
Hi, Is it possible to configure asterisk so it doesn't play MoH from certain phones? Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Wanpipe installation problems
Rory Campbell-Lange wrote: I'm trying to install wanpipe on my new 2.6.21-2-amd64 core 2 duo machine (Debian's amd64 works with EMT64 too) to run Asterisk. I'm getting compilation errors when trying to install the wanpipe utilities. Sangoma says that 2.6.21/22 is not supported yet, just 2.6.20. They're working on it. - Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why is nonce=584760da used in sip packets?
well in this case im using asterisk as a client(UA) to connect to my other asterisk server. So is this a bug in asterisk. im using asterisk 1.4.2 both as a client and server. On 8/15/07, Watkins, Bradley [EMAIL PROTECTED] wrote: You have on your hands a broken UA, since it is not responding to the changing nonce value. - Brad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham Sent: Wednesday, August 15, 2007 7:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] why is nonce=584760da used in sip packets? --- Transmitting (NAT) to 208.120.167.146:80 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 208.120.167.146:80;branch=z9hG4bK722c974c;received=208.120.167.146 From: sip:[EMAIL PROTECTED];tag=as1acc7245 To: sip:[EMAIL PROTECTED];tag=as1d329593 Call-ID: [EMAIL PROTECTED] CSeq: 19710 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=4f90fab4, stale=true Content-Length: 0 this stale=true field causes the asterisk server to display the following NOTICE on the cli NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce received from 'sip:[EMAIL PROTECTED] mailto:sip:[EMAIL PROTECTED] ' and this will continue happening unless the next register request uses the nonce field recieved in latest unauthorisation response from server, and untill then the user agent will not be able to register with the server. This will cause problems in our services. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disable MoH for certain phones
You can define a new class in musiconhold.conf with an empty directory. Create a directory with nothing in it /var/lib/asterisk/moh/empty Add this class to musiconhold.conf [empty] mode=files directory=/var/lib/asterisk/moh/empty Then for the phone' entry in sip.conf add: musiconhold=empty Be sure to conect to the CLI and do sip reload On 8/15/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, Is it possible to configure asterisk so it doesn't play MoH from certain phones? Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] http://www.shift8.biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some advice
William McCloskey [EMAIL PROTECTED] writes: I need a quick bit of advice from the list. We purchased an asterisk based phone system back about 6 months ago and we are using Cisco 7940G phones (I know, not everyone's favorites). We are using the second line on the phones for paging with a auto-answer, now my question is having the system call 20 of these paging extensions, should that be enough load to cause instability in the system? Our vendor is claiming it is causing the problems we are having, and I really find that hard to believe. Thoughts? Should that be enough to cause major stability problems? It's an Athlon 3800+ with 512mb ram and a Sangoma card with one PRI. Total of about 60 extensions (40 phones) on the system but only about 2-4 active calls at any given time with very little transcoding or other such intensive processes going on. Thanks, William You said that you only have around 2-4 active calls at any one time, but when the system is paging out what does your call count look like? If you are doing a simultaneous call to 20 different people that would shoot the load up on your server. I would do some testing to figure out if the instability occurs after the page goes out. Also, make sure that the pages are being disconnected (so you don't have lingering channels being used) and that you aren't doing *any* transcoding on the page out. 'show channels' 'uptime' :) -- Kyle Sexton ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote extension search?
Gordon Henderson [EMAIL PROTECTED] writes: On Tue, 14 Aug 2007, Nicholas Blasgen wrote: I've heard about this, but I really can't seem to find anything on it. I've got a strange setup that exists only because of firewall issues, and everything about it seems fine. The setup: SIP clients - Asterisk (office) - IAX - Asterisk (colocation) - SIP PSTN Termination All the extensions I want to be able to dial are on the colocation box. What I'd really like is for the office asterisk box to forward all extension requests it doesn't know about to the colocation Asterisk box. I think this is refered to as Trunking. I only need to do this in a single direction, if that's any easier to setup. With so many ways to do something, who defines proper :) Just from reading the subject line of this email it seems like DUNDi was built for this. Basically DUNDi will do a lookup and find which server an extension lives on. -- Kyle Sexton ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR billsec greater than duration
Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF with Aastra
In sip.conf I set this option: maxexpiry=10800 (3 hours)... The phones re-register once an hour. On 8/14/07, Steve Langstaff [EMAIL PROTECTED] wrote: What did you change? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: 14 August 2007 20:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] BLF with Aastra Well that was it... it is no longer timing out. On 8/14/07, James FitzGibbon [EMAIL PROTECTED] wrote: On 8/14/07, Matt [EMAIL PROTECTED] wrote: I have a 536i expansion module attached to a 57i-CT. The BLF lights on the 536i will light up and work fine for a while... however after a bit they seem to loose their ability to see if someone is on a phone. They still work to dial, if I try to dial, however, they don't light up when someone makes a call, or if their phone rings. If I reboot the phone, the lights start working again (for a while). Does 'sip show subscriptions' indicate that the 57i is still subscribed to the extension for updates? If not, you might have to do a test with 'sip debug peer aastraname' to confirm that the subscription is being made properly on phone startup and not being removed by the phone in response to some state change. A quick glance at chan_sip.c indicates that if a user agent tries to subscribe with an expiry time greater than 'maxexpiry' from sip.conf (default 3600 seconds), the subscription expiry in Asterisk will be silently changed to whatever the allowed maximum is. So if the Aastra is trying to subscribe for say 3 hours and Asterisk doesn't allow subscriptions greater than one hour, then notify messages will stop being sent after one hour until the Aatra re-subscribes. I haven't delved in very deep, so I can't tell if the response to the UA indicates the actual expiry Asterisk used, but even so you'd have to be certain that the Aastra respects an expiry in the response that differs from what it asked for. When you're doing the debug (hopefully on a quiet system), watch the phone boot, then use 'sip show subscriptions' to get the call-id of the subscription. Then watch for console messages indicating that the call has been destroyed (which should come at the 1 hour mark or whatever time the Aastra used for it's subscription length. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2.24 installation
got it working... looks like the tar file is corrupted or something redownload it again and installed it. Thanks! On 8/15/07, Anthony Francis [EMAIL PROTECTED] wrote: looks broken, is there an apps dir in the source directory? Mark Quitoriano wrote: is there a new way to install asterisk? im using centos 4.5 and trying to install asterisk. when i do make clean and make install i get this error. # make clean --snip-- make[1]: Leaving directory `/usr/src/asterisk- 1.2.24/apps' make: *** codecs: No such file or directory. Stop. make: *** [clean] Error 1 --snip-- # make --snip-- make[1]: Leaving directory `/usr/src/asterisk-1.2.24/apps' make: *** codecs: No such file or directory. Stop. make: *** [depend] Error 1 --snip-- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Mark Quitoriano, CCNA Fan the flame... http://www.spreadfirefox.com/?q=user/registerr=19441 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID Error causes problems for Polycom phones
Hi everyone, I have been dealing with a certain issue with a particular customer site for months now. The problem occurs when there is an error with caller id as shown in the following: WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error on channel 'Zap/3-1' When this happens, it appears that the call still goes through as I can see the caller still navigating through the systems menus and dialplan by watching the CLI. The problem however is manifested with polycom 301's that are setup with the system. When a call comes in after receiving that particular caller id error, the polycoms, which are on a group ring by the way, will all ring but you cannot pickup the call. The Answer|Reject soft buttons display, but only the reject button works. Pressing the Answer button or picking up the handset does nothing. Since only the Reject button works someone has to go to each phone and hit the reject button (4 polycoms in this department) so the ringing will at least stop. It's been about 3 months tracking this problem down (even drove the 2.5 hours back and forth to replace the sangoma card to try to fix the problem) and the customer is about ready to have me pull the system because of it. I can easily reproduce the problem with Polycom phones (but not the actual error). Just issue a .call file using the local channel calling one number and having the call bridged to a polycom phone (at least 301's here): Action: Originate Channel: local/[EMAIL PROTECTED] Context: to_meetme Exten: s Priority: 1 Async: true The above will cause the polycom to exhibit the behavior mentioned above. However, sending a .call file like the following causes the phone to work as it should: Action: Originate Channel: local/[EMAIL PROTECTED] Context: to_meetme Exten: s Priority: 1 Variable: CALLERID(num)=1234|CALLERID(name)=Homey D Clown Async: true I also have tried this with Aastra, Grandstream and XLite soft phones and they do not exhibit the same behavior. Instead these other phones simply show the default caller id info as set in sip.conf and allow you to answer them. Any help or suggestions would be greatly appreciated. OS: CentOS 4 Asterisk: 1.2.17 Sangoma A200 with 2 fxo ports. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID Error causes problems for Polycom phones
Am Mittwoch, den 15.08.2007, 10:14 -0400 schrieb Lee Jenkins: Hi everyone, I have been dealing with a certain issue with a particular customer site for months now. The problem occurs when there is an error with caller id as shown in the following: WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error on channel 'Zap/3-1' When this happens, it appears that the call still goes through as I can see the caller still navigating through the systems menus and dialplan by watching the CLI. The problem however is manifested with polycom 301's that are setup with the system. When a call comes in after receiving that particular caller id error, the polycoms, which are on a group ring by the way, will all ring but you cannot pickup the call. The Answer|Reject soft buttons display, but only the reject button works. Pressing the Answer button or picking up the handset does nothing. To me this looks like a firmware problem in your phones. Perhaps a firmware update could fix this. However - as it looks to me - the firmware chokes on some CALLERID strings, not on others. What is the caller id that is displayed in the error case? Perhaps you could get around by having a dialplan hook that rewrites the callerid to 000 if that invalid callerid comes in. Maybe those phones just choke on CALLERIDs with empty num or name With your test .call file that reproduces the problem, if you insert a line in your dialplan before the Dial() happens, that reads Set(CALLERID(all)=000) does that help? Does Set(CALLERID(num)=000) alone help, does Set(CALLERID(name)=000) ? BR Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] slightly OT: Polycom SIP phones
How difficult is it to change the firmware load on a Polycom phone from MGCP to SIP? I have a number of 500/600 phones and see some used phones being offered with MGCP installed. I have the SIP firmware but have never had to migrate between content loads. Any expected gotcha's? Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Patent issues, what features we can't use?
Thanks Mark for your detailed email. I have no plan to hide. I am in business and am ready to face any challenges. Just wanted to know where I stand if it comes to deal with patent issues, because there are companies out there who are going thorough this issue once they started to make good profits. But thats right, one can' t do any business if he starts to worry about patent issues. Whatever you want to do, someone else has already patended it. You can always verify it on google patents search. On 8/15/07, Mark Coccimiglio [EMAIL PROTECTED] wrote: Zeeshan, First off, if your fear of being sued is what stops you from doing business then get out of the industry or get over it. Its a risk we all take everyday (not just in VoIP). You build up a core of Insurance and Defensive Patents to protect yourself. Risk is just part of doing business. Elements of the Asterisk that are clearly incompatible with the Dual License model are not included in the regular distribution. You may find them as add-on modules or in Trunk (If it supports a free development/education license) but not as a part of the regular distribution. To address the real issue... In the USA in recent years companies have been granted broadly worded patents. People at the patent office are clerks and not engineers. Plus, they have to deal with ALL INDUSTRY (e.g. Medical, Aviation, Computer Science, Earth Science, Early Childhood Development, Mining, Agriculture, Automotive, Maritime, Textile, Nuclear Physics, Beauticare, Electronics, Chemistry, Mechanics, Pharmaceutical, etc...etc...etc...) not just Telecom. It is quite literally impossible to understand enough about everything to make clear judgments as to what is truly patentable and what is not. The patent office position is basically Spell it out to us and let the courts figure out the rest. While most broadly worded patents are unenforceable it still takes a legal process to get the patents dismissed as too vague. That process can be VERY costly for the person sued as well as the suer (sp). For a large telecom is all just part of the cost of doing business. Most smaller companies (e.g. us guys) are forced to settle because we haven't the millions of dollars needed to defend ourselves. Now that being said where does the g729 patent (and the like) fit in? A patent like g729 is actually VERY specific about what it does and how to do it. Sure its a software patent but there is little room in the wording about what it accomplishes, by what means and the limitations of the patent. Plus the price is very reasonable at $10/channel (non-transcoding pass through requires no License). Additionally, g729 is not the only game in town when it comes to low-bandwidth codecs. (Personally I like to use g726-32 its lightweight and transcodes to/from uLaw easily...but I digress). This varies from some other software patents for One-Click-Checkout or Online Shopping Cart. They are both patented and every challenge has been settled out of court, thus they still stand a viable patents. Ultimately the question comes down to...Do you want to stay home and hide or would you rather come out and play? Just my input, Mark C. Zeeshan Zakaria wrote: Hi everybody, As the Asterisk community is getting larger and larger, I was wondering that the features which are provided in Asterisk and are programmed by the open source community under GPL, or GUIs like FreePBX which also come loaded with wonderful features and uses same Asterisk, are they anywhere violating any patent laws? Most of the features work the same way as Nortel, Avaya and other PBX systems. Is there anyone who owns these features and will come one day to claim his royalties? When I deploy an asterisk soultion for a customer, is there any violation of any patent or copyright laws anywhere? Of if I use my own Asterisk server to provide services to some customers, am I violating any patent laws by not paying the royalties to some patent owners? I heard people saying that IVR technology is patented and google search for patents also say so. But we all are using IVR for ourselves and our customers without paying royalties to anyone. But when it comes to using g729, all of a sudden royalty issue comes in. So what is right to use and what is not? -- Zeeshan A Zakaria -- As I slowly sip my coffee I feel my humanity start to slip back into me and realize what a foul beast humanity really is. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] iaxmodem, chan_capi, hylafax problem and faxing in general
Can anybody point me in the right direction please? I'm having some issues getting iaxmodem and hylafax to talk to each other. I have no doubt that someone has had this type of issue before but I can't find anything useful in the archives or on Google. Under RH9, with chan_capi 7.1, Asterisk 1.2.24, with an AVI Fritz ISDN2e BRI card all working perfectly... I've downloaded and installed iaxmodem (2 days ago -- latest version - I'm not able to check exact release from where I am right now). It runs OK, and registers with Asterisk. I've downloaded and installed hylafax (again 2 days ago -- latest version). It installs OK and runs. I copied the suggested config file from iaxmodem directory to /var/spool/hylafax/config/etc (unchanged) I run iaxmodem before starting hylafax, and a device /dev/ttyIAX appears. Running iaxmodem in non-daemon mode shows no errors. I've not modified the default iaxmodem config other than in terms of username and password etc to register with asterisk. The problem is that when I run faxstat, it does not show hylafax connected to any tty. And when I try to run faxaddmodem (just to see what might happen) and select ttyIAX, I get an error saying that hylafax can't detect the speed of the device and that I should set it manually, and then iaxmodem promptly crashes at that point. I've poked and prodded generally but I'm getting nowhere. Does anybody have any suggestions as to where to start looking for the cause of the problem? For that matter, is this the right way to go? The idea is to have a central fax server running on the Asterisk box, receiving and sending via the ISDN BRI line. Ideally I'd like fax to email facilities and some point, and the ability for clients across our network to send faxes. I've seen that there's also another option, Astrifax, but that seems to require a separate installation of spandsp, java, and other bits which seems like a bit of overkill. Pointers or suggestions would therefore be appreciated. Faris. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Callback DTMF Problem
Hello All, I don't understand where is the problem... I have Callback setup and it works fine when tested within US. Works fine meaning the DTMF tones are passed when prompted to enter the phone number. But when I test with some international countries, callback works but DTMF tones are not passed... Is it: - a) Asterisk problem? b) Callback problem? c) VoIP provider problem? Under sip.conf I have dtmfmode = auto. Please help... Cheers, Nitesh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Events
Hi All, Can anybody send me a complete list of sip events. i know only 3 of those whihc are register, message-summary, message notification. message-summary event is causing some problems actually. My client sends a bad-event response if it recieves a message-summary event in a NOTIFY sip packet. I have little knowledge of sip events. So if anybody knows a good link plz share. And if u know how to fix the bad event message then plz tell also. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Events
http://www.faqs.org/rfcs/rfc3261.html Rizwan Hisham wrote: Hi All, Can anybody send me a complete list of sip events. i know only 3 of those whihc are register, message-summary, message notification. message-summary event is causing some problems actually. My client sends a bad-event response if it recieves a message-summary event in a NOTIFY sip packet. I have little knowledge of sip events. So if anybody knows a good link plz share. And if u know how to fix the bad event message then plz tell also. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Events
At least with my Manager API, I have the ability to simply set a default event handler and using that I can dump all events as the pass though. Then I setup a case switch and act on the ones I want. But the manager events I like are LINKED and HANGUP. http://www.voip-info.org/wiki/view/asterisk+manager+events On 8/15/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi All, Can anybody send me a complete list of sip events. i know only 3 of those whihc are register, message-summary, message notification. message-summary event is causing some problems actually. My client sends a bad-event response if it recieves a message-summary event in a NOTIFY sip packet. I have little knowledge of sip events. So if anybody knows a good link plz share. And if u know how to fix the bad event message then plz tell also. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR billsec greater than duration
Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Events
Ah, I correct myself. I see, you wanted to know the headers for each SIP packet. Makes a lot more sense now. On 8/15/07, Anthony Francis [EMAIL PROTECTED] wrote: http://www.faqs.org/rfcs/rfc3261.html Rizwan Hisham wrote: Hi All, Can anybody send me a complete list of sip events. i know only 3 of those whihc are register, message-summary, message notification. message-summary event is causing some problems actually. My client sends a bad-event response if it recieves a message-summary event in a NOTIFY sip packet. I have little knowledge of sip events. So if anybody knows a good link plz share. And if u know how to fix the bad event message then plz tell also. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID Error causes problems for Polycom phones
Anselm Martin Hoffmeister wrote:
Am Mittwoch, den 15.08.2007, 10:14 -0400 schrieb Lee Jenkins:
Hi everyone,
I have been dealing with a certain issue with a particular customer site
for months now. The problem occurs when there is an error with caller
id as shown in the following:
WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error
on channel 'Zap/3-1'
When this happens, it appears that the call still goes through as I can
see the caller still navigating through the systems menus and dialplan
by watching the CLI.
The problem however is manifested with polycom 301's that are setup with
the system. When a call comes in after receiving that particular caller
id error, the polycoms, which are on a group ring by the way, will all
ring but you cannot pickup the call. The Answer|Reject soft buttons
display, but only the reject button works. Pressing the Answer button
or picking up the handset does nothing.
To me this looks like a firmware problem in your phones. Perhaps a
firmware update could fix this. However - as it looks to me - the
firmware chokes on some CALLERID strings, not on others. What is the
caller id that is displayed in the error case? Perhaps you could get
around by having a dialplan hook that rewrites the callerid to 000 if
that invalid callerid comes in. Maybe those phones just choke on
CALLERIDs with empty num or name With your test .call file that
reproduces the problem, if you insert a line in your dialplan before the
Dial() happens, that reads
Set(CALLERID(all)=000)
does that help? Does
Set(CALLERID(num)=000)
alone help, does
Set(CALLERID(name)=000)
?
BR
Anselm
Anselm,
Thanks for responding. My apologies as I should have mentioned that I
have tried several workarounds including the following test scripts I
placed on the server:
[check_time]
; -
; Called right after Answer() is called
; -
; check for default value in sip.conf
exten=s,1,GotoIf($[${CALLERID(name)} = UNKNOWN ]?set_no_callerid,s,1)
; check for null value
exten=s,2,GotoIf($[${CALLERID(name)} = ]?set_no_callerid,s,1)
exten=s,3,GotoIf($[${CALLERID(num)} = ]?set_no_callerid,s,1)
exten=s,4,Noop(CallerID: ${CALLERID(num)} ${CALLERID(name)})
exten=s,5,Set(FAIL_MENU=daytime|TIMEOUT_MENU=daytime)
exten=s,6,GotoIfTime(08:30-17:00,mon-fri,*,*,?daytime_ivr,s,1)
exten=s,7,Goto(after_hours,s,1)
[set_no_callerid]
exten=s,1,Set(CALLERID(num)=410555)
exten=s,2,Set(CALLERID(name)=UNKNOWN)
exten=s,3,Goto(check_time,s,1)
I'll update the firmware on the phones and see if that helps.
Thanks again,
--
Warm Regards,
Lee
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[asterisk-users] Dialplan / AGI autoanswer question
Hi. I've got a working dial plan on my home system but there are problems
with it and I was hoping someone more comfortable with dial plans might be
able to help. In a nutshell here's what I'm currently doing on an incoming
outside phone call
[default]
Set(TIMEOUT(digit)=3
Set(TIMEOUT(response)=60
exten = s,1,NoOp(Answering in default context)
exten = s,2,Wait(1)
; look up the callerid name and state in the database
exten = s,3,AGI(cid_fix.php)
; print callerid
exten = s,4,NoOp(${CALLERID(name)} ${CALLERID(number)} ${CALLSTATE}
${DATETIME})
; short circuit the sequence if this is a blacklisted or whitelisted entry
exten = s,5,GotoIf($[${CALLSTATE} = black]?blacklisted,s,1)
exten = s,6,GotoIf($[${CALLSTATE} = white]?50)
; send a sit tone if we don't have a callerid
;
exten = s,7,Zapateller(nocallerid)
; after a certain time I don't want phone calls
;
exten = s,8,GotoIfTime(${AFTER_HOURS}|*|*|*?voicemail,s,1)
; no callerid
;
exten = s,9,GotoIf($[${CALLERID(number)} = ]?voicemail,s,1)
; If 800 number
;
exten = s,10,GotoIf($[${CALLERID(number):0:3} = 866]?voicemail,s,1)
exten = s,11,GotoIf($[${CALLERID(number):0:3} = 877]?voicemail,s,1)
exten = s,12,GotoIf($[${CALLERID(number):0:3} = 888]?voicemail,s,1)
exten = s,13,GotoIf($[${CALLERID(number):0:3} = 800]?voicemail,s,1:50)
; ring the phones and go to voicemail if nobody answers
;
exten = s,50,Dial(${LOCAL_LINES},20,tT)
exten = s,51,Goto(voicemail,s,1)
The intent of this sequence is to take the incoming callerid, replace it if
known with something in the database, and branch on the state from the DB
and time of the day.
One issue I'm having is that the AGI call seems to cause it to answer the
line. Normally that wouldn't be an issue but that answer seems to trip up
a number of automated systems and some people in addition to leaving a half
second pause in the ringing. Is there a way I can do the call to my lookup
script without answering the line yet? Or is there a better way to do what
I'm trying to do?
I'm working with asterisk 1.4.10
Thanks
--
Matthew HarrellManagers are like cats in a litter
Bit Twiddlers, Inc. box. They're always rearranging
[EMAIL PROTECTED] trying to cover up what they've done
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Re: [asterisk-users] Dialplan / AGI autoanswer question
Matthew Harrell wrote: Hi. I've got a working dial plan on my home system but there are problems with it and I was hoping someone more comfortable with dial plans might be able to help. In a nutshell here's what I'm currently doing on an incoming outside phone call [default] Set(TIMEOUT(digit)=3 Set(TIMEOUT(response)=60 These are missing closing brackets for one thing... -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxmodem, chan_capi, hylafax problem and faxing in general
Faris Raouf wrote: The problem is that when I run faxstat, it does not show hylafax connected to any tty. You're probably not running faxgetty (and your later comments below confirm this...) And when I try to run faxaddmodem (just to see what might happen) and select ttyIAX, I get an error saying that hylafax can't detect the speed of the device and that I should set it manually, and then iaxmodem promptly crashes at that point. You're probably not using HylaFAX+. The hylafax.org releases kill iaxmodem when faxaddmodem is run. See: http://hylafax.sourceforge.net/ Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P FXO click sounds
Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the same jacks, and the line is crystal-clear. Below is my zapata.conf, if you guys have any ideas how I might resolve this, I'd appreciate it. I have installed from sources Asterisk 1.2.22 and zaptel-1.2.19 on a debian etch x86_64. [channels] language=es context=ent-4229 ;rxwink=300 usecallerid=yes hidecallerid=no ; Whether or not to enable call waiting on internal extensions ; With this set to 'yes', busy extensions will hear the call-waiting ; tone, and can use hook-flash to switch between callers. The Dial() ; app will not return the BUSY result for extensions. ; callwaiting=yes threewaycalling=yes transfer=yes ;canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=yes echotraining=128 relaxdtmf=yes rxgain=3.0 txgain=3.0 callgroup=1 pickupgroup=1 immediate=no ;busydetect=yes ;busycount=4 callprogress=no ;busypattern=500,500 ;answeronpolarityswitch=yes ;hanguponpolarityswitch=yes ;callprogress=yes faxdetect=incoming faxdetect=outgoing signalling=fxs_ks group=1 channel=1 signalling=fxs_ks group=2 channel=2; singalling=fxs_ks group=3 channel=3; ;singalling=fxs_ks ;group=1 ;channel=4 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxmodem, chan_capi, hylafax problem and faxing in general
Thanks Lee. No, I'm definitely not running faxgetty - I didn't realise I was supposed to :-( And no, I'm not using the + version. Back to square one. At least I'm going in the right direction again! Only I've sidetracked and am currently trying to use capi4hylafax instead of iaxmodem which seems to work wonderfully but I'm having some issues with root verses uucp permissions which is spoiling my fun. Anyway, thanks again! Faris. (please excuse my top posting) -Original Message- Faris Raouf wrote: The problem is that when I run faxstat, it does not show hylafax connected to any tty. You're probably not running faxgetty (and your later comments below confirm this...) And when I try to run faxaddmodem (just to see what might happen) and select ttyIAX, I get an error saying that hylafax can't detect the speed of the device and that I should set it manually, and then iaxmodem promptly crashes at that point. You're probably not using HylaFAX+. The hylafax.org releases kill iaxmodem when faxaddmodem is run. See: http://hylafax.sourceforge.net/ Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iaxtel
Is iaxtel still around? I was not able to go to www.iaxtel.com . did the address changed? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] State of the Union: Vonage? Skype?
I've looked around a bit, and I'm still not sure I quite know what the state of the union is with regard to configuring SkypeIn/Out and Vonage services as trunk-side appearances on an Asterisk PBX? Any good clear pointers? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan / AGI autoanswer question
So besides the missing ) on line 1, I have some other comments: 1) You should replace your priority numbers with 'n'. Just so much easier to know that the issue isn't with priority numbers. And typing 'dialplan show context' is a nice way to see if everything is setup correctly. The 'n' is a personal choice, but the longer your application the better. 2) I thought I read somewhere that AGI was now auto-answering the channel. But I guess that's not right. AGI will auto-answer the channel if something causes it to do so. If you want to post your AGI code without any database commands, I'll glance at it. humm, I guess that's all I see. Everything else seems fine. It may be good to check the ChannelStatus once in a while just to debug where the channel is getting answered. http://gundy.org/asterisk/agi.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Client-negotiated Codec Instead of Transcoding?
Is there a way for voice media clients (like SIP phones and POTS/PSTN phones) that connect their call legs to Asterisk to negotiate a common codec that they both use at their end, so Asterisk doesn't have to transcode? Asterisk would know which codecs each client can use, and which each prefers, then find the one they each have in common so the fewest legs need Asterisk to transcode to their odd man out codec. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR billsec greater than duration
What is the definition of billsec, just out of curiosity? Seconds since the 200 OK from both ends / presumed media start? On Thu, 16 Aug 2007, Jaswinder Singh wrote: I made same thread few months ago and many people said that they dont have such records in plain asterisk install ( no freepbx ) . I was also using freepbx when i had this problem . Heres mine : mysql select count(*) from cdr where billsec duration; +--+ | count(*) | +--+ | 124 | +--+ this is out of 1749216 cdr records . I am also using freepbx btw . In all such cdr's duration is always 0 but billsec varies . On 15/08/07, Edoardo Serra [EMAIL PROTECTED] wrote: Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan / AGI autoanswer question
Set(TIMEOUT(digit)=3 Set(TIMEOUT(response)=60 These are missing closing brackets for one thing... That's just a cut and paste error. The real one has ending parens along with some other stuff outside of default mode. Didn't even notice that when I filled in the values -- Matthew Harrell Life is like a diaper - Bit Twiddlers, Inc. short and loaded. [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR billsec greater than duration
I made same thread few months ago and many people said that they dont have such records in plain asterisk install ( no freepbx ) . I was also using freepbx when i had this problem . Heres mine : mysql select count(*) from cdr where billsec duration; +--+ | count(*) | +--+ | 124 | +--+ this is out of 1749216 cdr records . I am also using freepbx btw . In all such cdr's duration is always 0 but billsec varies . On 15/08/07, Edoardo Serra [EMAIL PROTECTED] wrote: Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] State of the Union: Vonage? Skype?
Hi Jay, Skype can be used successfully with the ChanSkype module on supported platforms (Fedora Core 3, 4 or 5 or Ubuntu 6.04). It's $19USD for a single personal license, and tends to work quite well. It's not the easiest item to setup (the OS needs a window manager running on it, and each Skype channel require it's own user with it's own desktop session running for that user), but once you get it going I've rarely found it to fail. I used it with the unlimited outgoing calling to North America from Skype, and it's saved me quite a bit. AR On 8/15/07, Jay R. Ashworth [EMAIL PROTECTED] wrote: I've looked around a bit, and I'm still not sure I quite know what the state of the union is with regard to configuring SkypeIn/Out and Vonage services as trunk-side appearances on an Asterisk PBX? Any good clear pointers? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 3-com Model 3102 IP-Phone / Sip firmware download ?
3-com Model 3102 IP-Phone / Sip firmware download ? Has anyone every accomplished such? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load balancing SIP trunks?
I have 10 SIP trunks that I'd really like to round-robin load balance. Currently I have a macro that switches between available lines, but there really must be a function in Asterisk to do this on its own. So my question is just that, are there any easy ways for Asterisk to either balance between SIP trunks or even just a built in function to find the next available SIP trunk. I think using a macro to test the state of each trunk is silly, but it's the only method I've found. -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk SNOM Page - SNOM beeps intermittently
I have been trying to track this down for a while to no avail.
I have a variety of different SNOM phones (the entire 3XX series) and have
also tried on a variety of different Asterisk versions (pretty much the
whole 1.2 and 1.4 train)
When I Page() phones in Asterisk, I only intermittently get an audio
indication on the recipient phone. Audio Indicator is turned on, and I'm
using the basic/standard headers to page the phones.
exten = _83XXX,1,ChanIsAvail(SIP/${EXTEN:1}|js)
exten = _83XXX,n,SIPAddHeader(Call-Info: sip:xx.xx.xx.xx\;answer-after=0)
exten = _83XXX,n,Page(SIP/${EXTEN:1},d)
Occasionally the SNOM phones will actually beep before they pickup, but the
vast majority of the time this is not the case.
WRT to the SNOM phones, I've run (tested) everything from 5.x to 6.5.10 with
the same (intermittent) results.
Has anybody experienced this, and hopefully resolved?
Thanks,
--
Anthony Cennami
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Re: [asterisk-users] Dialplan / AGI autoanswer question
1) You should replace your priority numbers with 'n'. Just so much easier
to know that the issue isn't with priority numbers. And typing 'dialplan
show context' is a nice way to see if everything is setup correctly. The
'n' is a personal choice, but the longer your application the better.
I didn't know that was a feature until I just checked it out on the wiki but
I'm throwing it in. Thanks for the update
2) I thought I read somewhere that AGI was now auto-answering the channel.
But I guess that's not right. AGI will auto-answer the channel if something
causes it to do so. If you want to post your AGI code without any database
commands, I'll glance at it.
Well the issue may have resolved itself although I don't exactly know what
caused it to. I'll experiment more to find out since it must have been
something I changed earlier today
It may be good to check the ChannelStatus once in a while just to debug
where the channel is getting answered.
http://gundy.org/asterisk/agi.html
Also useful, thanks. I may have another use for that if I can check on the
status of remote Zap channels on my other server
If you don't mind I have two other things related to this setup that I
remembered when I was fiddling with it right now.
First, it seems I have to have a 2 - 3 second wait before the AGI call in
order to get valid CID data. Usually 2 seconds suffices for this one setup
but during that time the caller has had two rings before the local extension
has even begun to ring. Is there something I am doing wrong that causes it
to take so long to get the CID?
Finally, if I call from a remote site, it goes to voicemail, I hang up
before leaving a message, and then quickly call right back again I get what
sounds like a fax tone. I'm not specifying anything about faxes in the
extensions.conf and zapata.conf has all the fax stuff commented out. My
voicemail extension looks like
[voicemail]
exten = s,1,NoOp(${CALLERID(number)} going to voicemail)
exten = s,2,Set(vmnum=${VMCOUNT(${MAILBOX})})
exten = s,3,Background(mtn/greeting)
exten = s,4,Voicemail(s${MAILBOX})
exten = s,5,GotoIf($[${VMCOUNT(${MAILBOX})} = ${vmnum}]?7)
exten = s,6,NoOp(${CALLERID(number)} left a voicemail - sending alert)
exten = s,7,Hangup
Any thoughts on how I can stop whatever this fax issue is?
--
Matthew Harrell Think of it as evolution in action
Bit Twiddlers, Inc.
[EMAIL PROTECTED]
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Re: [asterisk-users] TDM400P FXO click sounds
[EMAIL PROTECTED] wrote: Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the same jacks, and the line is crystal-clear. Below is my zapata.conf, if you guys have any ideas how I might resolve this, I'd appreciate it. I have installed from sources Asterisk 1.2.22 and zaptel-1.2.19 on a debian etch x86_64. Have you tried running fxotune on it? If you have (or haven't) make sure you try the zaptel-1.4 version of fxotune. It has improved significantly since 1.2. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Did not work either...Thank you! Otis Michiel van Baak wrote: On 15:02, Sun 12 Aug 07, OCOSA ListAcct wrote: exten=5,2,Dial(SIP/supportSIP/support2,2,tr) Make this line read: exten=5,2,Dial(SIP/supportSIP/support2,,tr) That should do the trick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GUI for Asterisk realtime
Are there any nice GUIs out there for Asterisk Realtime? Google doesn't yield much. I spent a day trying to get VoiceOne to work without much success. Thanks, Mike Clark ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan / AGI autoanswer question
First, it seems I have to have a 2 - 3 second wait before the AGI call in order to get valid CID data. Usually 2 seconds suffices for this one setup but during that time the caller has had two rings before the local extension has even begun to ring. Is there something I am doing wrong that causes it to take so long to get the CID? Check out the CDR configuration. I do my CDR via MySQL and I don't think that does buffering, but I know for sure the normal CSV format (and standard configuration file) has options for buffering before saving. I can't really think how that would change recieving the CDR information during a call, but it's possible something along those lines. I'm making it up really since the more I think about it the less that sounds possible. But search the documentation on the CDR or maybe ask the Asterisk Dev group about when the CDR fields get filled. It might even be possible that CDR infromation isn't accessable untill the line has been answered and that's the delay. Finally, if I call from a remote site, it goes to voicemail, I hang up before leaving a message, and then quickly call right back again I get what sounds like a fax tone. I'm not specifying anything about faxes in the extensions.conf and zapata.conf has all the fax stuff commented out. My voicemail extension looks like What is the channel type to the remote site? IAX, SIP, analog? I've seen alarm systems that pick up analog lines they're attached to if thre are back to back calls. That's the best I can come up with there. Maybe someone else has a better idea. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan / AGI autoanswer question
First, it seems I have to have a 2 - 3 second wait before the AGI call in order to get valid CID data. Usually 2 seconds suffices for this one setup but during that time the caller has had two rings before the local extension has even begun to ring. Is there something I am doing wrong that causes it to take so long to get the CID? CallerID info is sent between the first and second ring. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan / AGI autoanswer question
On Wed, 15 Aug 2007, Matthew Harrell wrote:
The intent of this sequence is to take the incoming callerid, replace it if
known with something in the database, and branch on the state from the DB
and time of the day.
FWIW: I do something similar, but purely in dial-plan using the astdb -
here's an extract, it demos jumping to named labels too:
exten = incoming,1,Noop(New incoming call. CallerId is ${CALLERID(all)})
; See if we have a name:
exten = incoming,n,GotoIf($[${CALLERID(name)} != ]?gotName)
; OK. No Name. Set a default name
exten = incoming,n,Set(CALLERID(name)=Unknown)
exten = incoming,n(gotName),Noop(Carrying on after name check)
; See if we have a number:
exten = incoming,n,GotoIf($[${CALLERID(number)} != ]?gotNumber)
; OK. No Number. Set a default number
exten = incoming,n,Set(CALLERID(number)=Withheld)
exten = incoming,n(gotNumber),Noop(Carrying on after number check)
; Now see if the number is the our internal database which will override any
; name we might have.
exten = incoming,n,Set(name=${DB(cid/${CALLERID(number)})})
exten = incoming,n,GotoIf($[${name} = ]?doneCIDprocessing)
exten = incoming,n,Set(CALLERID(name)=${name})
exten = incoming,n,Noop(We set our name to ${name} from the database)
exten = incoming,n(doneCIDprocessing),Noop(Done with incoming CID processing -
we have a call from ${CALLERID(all)})
You could keep a separate list of number states in the database too, and
extract this. Eg. above the line where we get the name out of the astdb
above:
exten = incoming,n,Set(CALLSTATE=${DB(state/${CALLERID(number)})})
and so on...
I'm not sure what (if any!) benefit this might have over running an
external PHP application... I'd like to think it might actually be quicker
for simple cases like this, but I've never benchmarked it.
Gordon
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Re: [asterisk-users] Dialplan / AGI autoanswer question
First, it seems I have to have a 2 - 3 second wait before the AGI call in order to get valid CID data. Usually 2 seconds suffices for this one setup but during that time the caller has had two rings before the local extension has even begun to ring. Is there something I am doing wrong that causes it to take so long to get the CID? CallerID info is sent between the first and second ring. Well that would explain that problem, wouldn't it? Is there a proper way to wait for the CID data to be filled in if available or is Wait(2) my best option? -- Matthew Harrell Nondeterminism means never Bit Twiddlers, Inc. having to say you are wrong. [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan / AGI autoanswer question
: Check out the CDR configuration. I do my CDR via MySQL and I don't think : that does buffering, but I know for sure the normal CSV format (and standard : configuration file) has options for buffering before saving. I can't really : think how that would change recieving the CDR information during a call, but : it's possible something along those lines. I'm making it up really since : the more I think about it the less that sounds possible. But search the : documentation on the CDR or maybe ask the Asterisk Dev group about when the : CDR fields get filled. It might even be possible that CDR infromation isn't : accessable untill the line has been answered and that's the delay. Do you mean CDR? I was asking about the delay in the CID (caller id) values and I'm just not sure whether you read that as CDR or whether the CDR settings would really affect the CID values : What is the channel type to the remote site? IAX, SIP, analog? I've seen : alarm systems that pick up analog lines they're attached to if thre are back : to back calls. That's the best I can come up with there. Maybe someone : else has a better idea. This happens on an analog incoming call through a Zap channel. I haven't noticed the same problem on the SIP or IAX channels but I haven't really tried to recreate it either -- Matthew Harrell Never raise your hand to your Bit Twiddlers, Inc. children - it leaves your [EMAIL PROTECTED] midsection unprotected. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan / AGI autoanswer question
Wait(2) is what I do. Matthew Harrell wrote: First, it seems I have to have a 2 - 3 second wait before the AGI call in order to get valid CID data. Usually 2 seconds suffices for this one setup but during that time the caller has had two rings before the local extension has even begun to ring. Is there something I am doing wrong that causes it to take so long to get the CID? CallerID info is sent between the first and second ring. Well that would explain that problem, wouldn't it? Is there a proper way to wait for the CID data to be filled in if available or is Wait(2) my best option? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan / AGI autoanswer question
Thanks. I was hoping there might be a way to detect whether the CID routine was done or not. I've still seen occasions where it wasn't available for callers that I know had it. Maybe my phone service is just a little slow sometimes Wait(2) is what I do. Matthew Harrell wrote: First, it seems I have to have a 2 - 3 second wait before the AGI call in order to get valid CID data. Usually 2 seconds suffices for this one setup but during that time the caller has had two rings before the local extension has even begun to ring. Is there something I am doing wrong that causes it to take so long to get the CID? CallerID info is sent between the first and second ring. Well that would explain that problem, wouldn't it? Is there a proper way to wait for the CID data to be filled in if available or is Wait(2) my best option? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Harrell Never underestimate the power of Bit Twiddlers, Inc. very stupid people in large groups. [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some advice
William McCloskey wrote: The stability problems we have seem to be related to asterisk crashing the apache install on the box when the PHP scripts are performing functions via asterisk. Don't know exactly how they work it all, but that's the gist of it. Are the PHP scripts connected with paging at all? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
Try some of these suggestions. http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 15, 2007 12:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM400P FXO click sounds Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the same jacks, and the line is crystal-clear. Below is my zapata.conf, if you guys have any ideas how I might resolve this, I'd appreciate it. I have installed from sources Asterisk 1.2.22 and zaptel-1.2.19 on a debian etch x86_64. [channels] language=es context=ent-4229 ;rxwink=300 usecallerid=yes hidecallerid=no ; Whether or not to enable call waiting on internal extensions ; With this set to 'yes', busy extensions will hear the call-waiting ; tone, and can use hook-flash to switch between callers. The Dial() ; app will not return the BUSY result for extensions. ; callwaiting=yes threewaycalling=yes transfer=yes ;canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=yes echotraining=128 relaxdtmf=yes rxgain=3.0 txgain=3.0 callgroup=1 pickupgroup=1 immediate=no ;busydetect=yes ;busycount=4 callprogress=no ;busypattern=500,500 ;answeronpolarityswitch=yes ;hanguponpolarityswitch=yes ;callprogress=yes faxdetect=incoming faxdetect=outgoing signalling=fxs_ks group=1 channel=1 signalling=fxs_ks group=2 channel=2; singalling=fxs_ks group=3 channel=3; ;singalling=fxs_ks ;group=1 ;channel=4 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Wanpipe installation problems
Dr. Michael J. Chudobiak wrote: Rory Campbell-Lange wrote: I'm trying to install wanpipe on my new 2.6.21-2-amd64 core 2 duo machine (Debian's amd64 works with EMT64 too) to run Asterisk. I'm getting compilation errors when trying to install the wanpipe utilities. Sangoma says that 2.6.21/22 is not supported yet, just 2.6.20. They're working on it. In the interim the beta drivers will work. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
Please explain to me how FXO tune would fix popping and clicking sounds??? -Original Message- From: Matthew Fredrickson [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 15, 2007 2:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TDM400P FXO click sounds [EMAIL PROTECTED] wrote: Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the same jacks, and the line is crystal-clear. Below is my zapata.conf, if you guys have any ideas how I might resolve this, I'd appreciate it. I have installed from sources Asterisk 1.2.22 and zaptel-1.2.19 on a debian etch x86_64. Have you tried running fxotune on it? If you have (or haven't) make sure you try the zaptel-1.4 version of fxotune. It has improved significantly since 1.2. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
shadowym wrote: Please explain to me how FXO tune would fix popping and clicking sounds??? If they are caused by a poorly-tuned echo canceller. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Wanpipe installation problems
On 15/08/07, Stephen Bosch ([EMAIL PROTECTED]) wrote: Dr. Michael J. Chudobiak wrote: Rory Campbell-Lange wrote: I'm trying to install wanpipe on my new 2.6.21-2-amd64 core 2 duo machine (Debian's amd64 works with EMT64 too) to run Asterisk. I'm getting compilation errors when trying to install the wanpipe utilities. Sangoma says that 2.6.21/22 is not supported yet, just 2.6.20. They're working on it. In the interim the beta drivers will work. I've been in contact with Sangoma support and have done a lot of re-compiling in the last 24 hours. Stick to .20 or earlier kernels, even for the beta version. We reverted to Debian stable (.18 kernel) for our installation platform and all went very smoothly. Regards Rory -- Rory Campbell-Lange Director Campbell-Lange Workshop Ltd. [EMAIL PROTECTED] www.campbell-lange.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel update locks up computer from 1.2.9.1 to 1.2.19
I am trying to update a machine with a TE210P card setup as PRI. Running Centos 4.4. I stop asterisk, I do service zaptel stop. I look at lsmod and all zaptel modules are unloaded. I compile zaptel 1.2.19, I install zaptel. when I do the service zaptel start, the machine locks up. I reboot the machine and it locks up when loading zaptel. I have to shutdown the machine, take out the card, recompile the old driver, and reinstall it, shutdown the machine, reinstall the card, then reboot. The machine boots and its working again. What is going on here??? How can I get it to update to the new version? Jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Seeking opinions: Polycom IP330 phones?
Does anyone online have an opinion on these? I've used 500/510/6001/601 models before. Need to know if these apparently lesser models can be provisioned in the same way. Are end uers happy with them? Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] State of the Union: Vonage? Skype?
One short warning - I know of a company that tried to setup a larger installation of ChanSkype, and it didn't work very well at all. (huge memory use, crashes, lockups) PaulH On Wed, 2007-08-15 at 16:15 -0400, Alex Robar wrote: Hi Jay, Skype can be used successfully with the ChanSkype module on supported platforms (Fedora Core 3, 4 or 5 or Ubuntu 6.04). It's $19USD for a single personal license, and tends to work quite well. It's not the easiest item to setup (the OS needs a window manager running on it, and each Skype channel require it's own user with it's own desktop session running for that user), but once you get it going I've rarely found it to fail. I used it with the unlimited outgoing calling to North America from Skype, and it's saved me quite a bit. AR On 8/15/07, Jay R. Ashworth [EMAIL PROTECTED] wrote: I've looked around a bit, and I'm still not sure I quite know what the state of the union is with regard to configuring SkypeIn/Out and Vonage services as trunk-side appearances on an Asterisk PBX? Any good clear pointers? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seeking opinions: Polycom IP330 phones?
I did some testing on the 430, and was pretty happy with it. Paul Hales AsteriskIT On Wed, 2007-08-15 at 20:52 -0500, Michael Graves wrote: Does anyone online have an opinion on these? I've used 500/510/6001/601 models before. Need to know if these apparently lesser models can be provisioned in the same way. Are end uers happy with them? Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seeking opinions: Polycom IP330 phones?
I have deployed a couple 330's and they will use the same provisioning methods as the earlier models from Polycom, you just need to be sure and have the firmawre and configs for that version. The other thing I ran into, is the 330 did not ship with a power supply so you either go POE or buy the adapter in 5 packs. That may have changed, I order them the month they began shipping. I have not heard any complaints about them, but I deployed them in copy rooms and such not at anyone's desk. On 8/15/07, Michael Graves [EMAIL PROTECTED] wrote: Does anyone online have an opinion on these? I've used 500/510/6001/601 models before. Need to know if these apparently lesser models can be provisioned in the same way. Are end uers happy with them? Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel update locks up computer from 1.2.9.1 to 1.2.19
On Wed, Aug 15, 2007 at 09:22:45PM -0400, Jerry Geis wrote: I am trying to update a machine with a TE210P card setup as PRI. Running Centos 4.4. What is the output of: uname -r I stop asterisk, I do service zaptel stop. I look at lsmod and all zaptel modules are unloaded. I compile zaptel 1.2.19, I install zaptel. when I do the service zaptel start, the machine locks up. I reboot the machine and it locks up when loading zaptel. When exactly? If you manually run: modprobe zaptel # is that the right driver? insmod wct4xxp # you may need to wait for a while here. udevd takes its time on centos4 # to populate /dev/zap ztcfg where exactly will it lock up? Does it print anything to the console? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk multiport
hot to asterisk multiport...??? example 5060, 5061, 5080 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk multiport
Off the cuff, I can't recall if asterisk can listen for (in this case I assume) SIP on multiple ports. It would be quite easy to do this redirection with iptables, though. On 8/15/07, Walter Willis [EMAIL PROTECTED] wrote: hot to asterisk multiport...??? example 5060, 5061, 5080 -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mitel IP 5020 phones
Hi, folks: I've come into some Mitel 5020 IP phones. A client has made a significant investment in them and we want to see if we can use them in a new system. Are these even SIP sets? I haven't been able to find out. Mitel's site barely covers them (I was only able to find some user guides, which are effectively useless; they say nothing about configuration and provisioning). Has anybody used these with Asterisk? Any other feedback or advice out there? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
