RE: [Asterisk-Users] Pause during dial
Henry Devito:
exten = 3,8,Dial(sip/${destination}D{$pin})
^^
Awoogah. Awoogah.
Nick.
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[Asterisk-Users] Frequency Shift
Title: Frequency Shift Hello, I am using * as a SIP proxy with several SIP clients. The SIP clients are SJPhone Soft phones. All clients are inside a firewall and the Server is inside too. All is working fine, but the speech sounds like Micky Mouse. If you feed one client´s (Mic) input with a permanent tone i.e. a 440 Hz Sinus wave it´s frequency on the (Speaker) output of the client you are connected to is shifted to a higher frequency. In addition to this you can hear drop outs. Obviously the samling rate on the sender´s side does not fit the receivers rate. I do not understand this because both phones are using G.711 ALAW. Taking a look at *´s channels with the help of it´s command line interface shows ALAW for both channels too. I set reinvite=no in the sip.conf file, because SJPhone did not support this and the connection broke down. So if I understand things right the conversion error could also be caused by *, because it stays inside the rtp connection. Does anybody know something about this phenomena?? Thanks in advance joerg. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream BugeTone 101 - Multi-Server setup ???
I am reading the manual from Bugetone 101 and found on page 19, the setting for [8] SIP SP-1 till SP-9 That would be nice! Could leave the FWD number in place, while I test my Asterisk setup !! However, I did not find out how I can setup SP-2 ~ SP-9 (Only configured SIP server(s) are displayed) I am not at the phone, bye Ronald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P / FXO / Polarity Reversal
Hi there, I'm trying to set up a small asterisk box for our company, and am using a TDM400P with an FXO module in it for one of the external PSTN lines. I'm having problems getting Asterisk to detect the remote caller hangup; when a call is received, I get the following messages on the Asterisk console: Exception on 16, channel 2 Got event Polarity Reversal(17) on channel 2 (index 0) Dunno what to do with event 17 on channel 2 and another of these when the caller hangs up. As far as I can tell, the first could be an indication that caller id is present, although I am getting that through ok. Most of the information I can find seems to suggest that Asterisk is aware of 'Open Loop Disconnect' and the Asterisk Wiki page on Disconnect Supervision does seem to indicate that reversing the polarity is one way of doing Disconnect Supervision. While I go off and try to find the relevant bits of code, I thought I'd just post a query here to see if anyone can quickly point me at the approrpiate configuration settings etc. Many regards marty -- Marty Lee e: [EMAIL PROTECTED] Technical Directorm: +44 7747 567 267 Upstart Training Ltd f: +44 871 433 8922 Scotland, UK w: http://www.upstart-training.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No SIP registration but user has dialled out?!?
hi when looking into the sipfriends table (using mysql sipfriends from asterisk cvs version -r v1-0), I see timestamp and ipaddr set to 0/NULL. When looking into the CDR, the user has dialled out recently. Also sip show peer xxx shows no data. How can this be true? roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Top posting
I must admit I live in perpetual fear of forgetting to switch of html or rtf formatting (useful for work) and top posting. I can understand the issue with the former but can see absolutely no reason why top posting is such a problem. In fact I'd far prefer it. I get to my e-mail in batches and bottom posting means I've got to wade through stuff I've just read. I totally agree with snipping extensively. So that I can understand the almost religious fervour on this point could someone please explain to me why top posting is so hated!! I can understand that if you are responding to multiple points in an e-mail then you should reply below each point snipping out what is irrelevant to your reply in the original e-mail. If you're responding to an entire e-mail then the proper approach to my mind would to do as you would in business letters and start with a short paragraph explaining what you're doing (e.g. In response to Fred's e-mail about AMD MP motherboards and interrupts, I guess most of us are too lazy to do this so we just leave the original text in the e-mail. If we're really lazy we don't snip the irrelevant stuff out. Am I missing something totally?! I'm just about to go and get my flak jacket and helmet in anticipation of the responses. :) Regards, George ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra/Sayson 480i eval
On Wed, 2004-11-10 at 22:25 -0600, Rich Adamson wrote: Just a quick FYI for the Aastra/Sayson 480i SIP phone Just received one and now have it running with *. - Unit came with SIP v1.0.0.34 Release code 0035-00-00 installed. No CDROM shipped with the unit, and a quick look at www.aastra.com and www.sayson.com sites didn't appear as though one can download firmware upgrades. Not sure where one is supposed to get them. There is a little piece of paper that comes with the phone. There it says to contact Sayson support so they can assign you an account on the support site where the firmware is. New firmware is at version 1.0.0.41 There is still a very big problem with this phone, the dial plan will only allow you to dial 10 digits. For local numbers this is not a problem, but you cannot dial long distance. Carlos Chavez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quasi-skype channel for Asterisk?
Hi, (B (BDate: Thu, 11 Nov 2004 08:42:16 +0200 (SAST) [zone:-], [EMAIL PROTECTED] (Bmentioned in msg: Re: [Asterisk-Users] quasi-skype channel for Asterisk? (Bthat ... (B (B On Wed, 10 Nov 2004, Kuniyoshi Murata wrote: (B (B http://www.pcphoneline.com/skype (B (B If I have a spare PC-AT running Windows 2000/XP and use their devices to (B convert skype's input and output to conventional phone jack, I guess I can (B connect that to Asterisk and skype can be one of the channels. (B (B Is my understanding right? (B (B (B Yes, but you are still contravening Skype's terms and conditions of use. (B (B1. How and What clause of that is contravening to connecting to another PBX (Bincluding Asterisk? I roughly read through them but I couldn't find obvious (Bone. Could you specify the exact phrase? (B (B2. The vender of these Skype to FXO/FXS converter is suggesting in their web (Bsite that their devises enable users to connect Skype to PBX. Are you saying (Bthat they are not obeying to clauses of Skype rules? (B (B (B-- (BKuniyoshi Murata.iChat/AIM:macwebcaster (BEnglish-Japanese Interpreter mailto:[EMAIL PROTECTED] (BMacintosh Webcast Specialisthttp://www.macwebcaster.com (B (B (B (B___ (BAsterisk-Users mailing list ([EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Top posting
George Gardiner [EMAIL PROTECTED] writes: So that I can understand the almost religious fervour on this point could someone please explain to me why top posting is so hated!! Because there's such an enormous amount of communication one would like to take part in, and not enough time. The easier it is to quickly discover a) whether each item is interesting, and b) what is the exact context of the item, and of its constituent parts, the more interesting material we can actually read. Therefore, top posting and bottom posting are equally bad; the ideal is an easily readable text that's placed into its proper context by short quotes of the relevant bits of previous communication. (Note: *short* quotes. If the reader wants the full text of the previous message, retrieving that message takes but a moment, so there's no need to quote it all.) For my own part, I have taken to ignoring anything that is badly formatted, top posted, bottom posted, or otherwise makes it difficult to quickly get into the flow of the communication. My default is to move on; only if your posting quickly establishes that it is, in fact, interesting to me, will I read it. To put it bluntly: if you can't be bothered to make an effort to communicate, what you say can't be very important. ;-) -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Top posting
From: Tom Ivar Helbekkmo [mailto:[EMAIL PROTECTED] Sent: 11 November 2004 09:38 My default is to move on; only if your posting quickly establishes that it is, in fact, interesting to me, will I read it. To put it bluntly: if you can't be bothered to make an effort to communicate, what you say can't be very important. -- I certainly agree with your sentiments in a general mailing list sense! I am of the opinion that this mailing list should entirely be devoted to a Question and Answer style, and that is all. So you point about Top of Bottom posting being irrelevant rings true. If people wish to discuss open ended topics (What XYZ Phone is the best? or Is VOIP going to cure world hunger?) then another mailing list should maybe be started. Asterisk - Users Technical and Asterisk - Users Discussion maybe??? The problem with ignoring badly formatted replies is more often than not the person replying couldn't careless if you ignored the reply but the person asking the question will care, especially if the information being supplied could be improved by your input! So sometimes its worth being a little more forgiving, for example if the original poster did go to significant lengths to provide a good question. I think the catch phrase should be Ask good questions and Give good answers. This includes all the things you mention. It would be extremely helpful if everytime someone gets an answer to their question as a way of thanks and etiquette they take it upon themselves to ensure that this answer is now covered in the WIKI. If this always happened and if people checked the WIKI the volume of repeat mails would drop hugely. For Example: Original Poster --- Asks Question LOOP Reply --- Request Improved Question (more detail / config files / logs /etc) Original Poster --- Resubmits Improved Question (Snipping irrelevant info) /LOOP Reply --- Answer Original Poster -- Reformats entire thread based on all answers and ensures question and answer are covered in an intuitive section of the WIKI Just my opinion. Alex This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't compile app_conference
Hi Henry, I have found your message in the mailing list archive from October where you describes compiling problem with the app_conference. Now I have exactly the same problem with it. Have you found any solution of this problem? Link to our message: http://lists.digium.com/pipermail/asterisk-users/2004-October/067961.html Thanx Rastislav ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Top posting
Oh, that's a great idea, Tom. Let's have everyone operate to your exacting standards. I can appreciate that not everyone did their degree in mail list etiquette and have lives to live and so want to be economical with their time. So for my part I scan emails top, bottom or otherwise posted and reply if I think I have a contribution to make or something to learn (in my experience knowledgeable people are often extremely busy and brief). Clearly if something has become illegible or doesn't include relevant information, it's not going to garner any attention or convey any useful information. But in my experience most posts on this list are good enough and some tolerance goes a long way. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Ivar Helbekkmo Sent: November 11, 2004 9:38 AM To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Top posting George Gardiner [EMAIL PROTECTED] writes: So that I can understand the almost religious fervour on this point could someone please explain to me why top posting is so hated!! Because there's such an enormous amount of communication one would like to take part in, and not enough time. The easier it is to quickly discover a) whether each item is interesting, and b) what is the exact context of the item, and of its constituent parts, the more interesting material we can actually read. Therefore, top posting and bottom posting are equally bad; the ideal is an easily readable text that's placed into its proper context by short quotes of the relevant bits of previous communication. (Note: *short* quotes. If the reader wants the full text of the previous message, retrieving that message takes but a moment, so there's no need to quote it all.) For my own part, I have taken to ignoring anything that is badly formatted, top posted, bottom posted, or otherwise makes it difficult to quickly get into the flow of the communication. My default is to move on; only if your posting quickly establishes that it is, in fact, interesting to me, will I read it. To put it bluntly: if you can't be bothered to make an effort to communicate, what you say can't be very important. ;-) -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk support for ISDN 1TR6 ?
hi, can someone give me any hints if the old german ISDN protocol '1TR6' is supported by asterisk. we have a potential customer who has an existing conventional PBX which has to be extended by an asterisk server. unfortunately this existing PBX speaks 1TR6 on it's ISDN ports. regards frank sautter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] xlite and asterisk
Hi, I havent received many replies so i was just wondering again if anyone has any thoughts of the 404 call not found issue.I have only a very basic configuration which can be seen below in the original email. I have since modified this so that each client (i.e. 2000 and 2001) have 'context=from-sip' included in their config and [from-sip] is in the extensions.conf file. I have now included the diagnostic log from the xlite client to see if that helps. Also when i do sip show peers I see: Username Host 2001 157.190.70.231 2000 84.203.148.14 The 84.203.148.14 is the address of asterisk, should the 2001 client be registering with that address too? Any help appreciated. Aisling. SEND TIME: 4679188 SEND 84.203.148.14:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 84.203.148.14:5061;rport;branch=z9hG4bKFDC50F318C6B44F09C2FCBBAE4DF34A 9 From: Aisling O' Driscoll sip:[EMAIL PROTECTED]:5061;tag=2391420445 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] CSeq: 6927 INVITE Proxy-Authorization: Digest username=2000,realm=asterisk,nonce=10dee878,response=fadc5f7ff0 2e3cbd5e8253d173f0b691,uri=sip:[EMAIL PROTECTED] Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 269 v=0 o=2000 4676744 4679018 IN IP4 84.203.148.14 s=X-Lite c=IN IP4 84.203.148.14 t=0 0 m=audio 8000 RTP/AVP 0 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE TIME: 4679298 RECEIVE 84.203.148.14:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 84.203.148.14:5061;branch=z9hG4bKFDC50F318C6B44F09C2FCBBAE4DF34A9;rece ived=84.203.148.14;rport=5061 From: Aisling O' Driscoll sip:[EMAIL PROTECTED]:5061;tag=2391420445 To: sip:[EMAIL PROTECTED];tag=as4ebaa89b Call-ID: [EMAIL PROTECTED] CSeq: 6927 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SEND TIME: 4679298 SEND 84.203.148.14:5060 ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 84.203.148.14:5061;rport;branch=z9hG4bKFDC50F318C6B44F09C2FCBBAE4DF34A 9 From: Aisling O' Driscoll sip:[EMAIL PROTECTED]:5061;tag=2391420445 To: sip:[EMAIL PROTECTED];tag=as4ebaa89b Contact: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] CSeq: 6927 ACK Max-Forwards: 70 Content-Length: 0 Original Message From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] xlite and asterisk Date: Wed, 10 Nov 2004 12:39:22 + 404 not found can mean many things, are you using a supporting codec? On Wednesday 10 November 2004 05:25 pm, Ashling O'Driscoll wrote: Hi, Hope somebody can help. I have two xlite clients that register with asterisk. They are called 2000 and 2001. 1)When 2000 rings 2001 a '404 not found' message is returned even though he is registered with asterisk. 2)When 2001 rings 2000, a 'call not approved' error is returned. I found a thread regarding the 'call not approved' error in the asterisk archives but no solution was posted. I have included the relevant portion of my config files below. If any further info is needed please let me know. Also how is it possible to dial a sip address e.g. sip:[EMAIL PROTECTED] from an xlite client? Thanks again, Aisling. sip.conf ;xlite client 1 [2000] type=friend username=2000 secret=whatever nat=yes host=dynamic mailbox=100 [2001] type=friend username=2001 secret=bla nat=yes host=dynamic mailbox=101 extensions.conf exten =3D 2000,1,Dial(SIP/2000,20) exten =3D 2001,1,Dial(SIP/2001,20) ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. - -- - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Heritage Communications Corporation Melbourne, FL USA 32935 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or
Re: [Asterisk-Users] xlite and asterisk
Hello, try this document (from the wiki): http://www.astmasters.net/stuff/X-Lite-and-Asterisk.pdf setting the auth param and the canreinvite and reinvite might help. -yair On Thu, 11 Nov 2004 10:38:55 -, Ashling O'Driscoll [EMAIL PROTECTED] wrote: Hi, I havent received many replies so i was just wondering again if anyone has any thoughts of the 404 call not found issue.I have only a very basic configuration which can be seen below in the original email. I have since modified this so that each client (i.e. 2000 and 2001) have 'context=from-sip' included in their config and [from-sip] is in the extensions.conf file. I have now included the diagnostic log from the xlite client to see if that helps. Also when i do sip show peers I see: Username Host 2001 157.190.70.231 2000 84.203.148.14 The 84.203.148.14 is the address of asterisk, should the 2001 client be registering with that address too? Any help appreciated. Aisling. SEND TIME: 4679188 SEND 84.203.148.14:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 84.203.148.14:5061;rport;branch=z9hG4bKFDC50F318C6B44F09C2FCBBAE4DF34A 9 From: Aisling O' Driscoll sip:[EMAIL PROTECTED]:5061;tag=2391420445 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] CSeq: 6927 INVITE Proxy-Authorization: Digest username=2000,realm=asterisk,nonce=10dee878,response=fadc5f7ff0 2e3cbd5e8253d173f0b691,uri=sip:[EMAIL PROTECTED] Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 269 v=0 o=2000 4676744 4679018 IN IP4 84.203.148.14 s=X-Lite c=IN IP4 84.203.148.14 t=0 0 m=audio 8000 RTP/AVP 0 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE TIME: 4679298 RECEIVE 84.203.148.14:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 84.203.148.14:5061;branch=z9hG4bKFDC50F318C6B44F09C2FCBBAE4DF34A9;rece ived=84.203.148.14;rport=5061 From: Aisling O' Driscoll sip:[EMAIL PROTECTED]:5061;tag=2391420445 To: sip:[EMAIL PROTECTED];tag=as4ebaa89b Call-ID: [EMAIL PROTECTED] CSeq: 6927 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SEND TIME: 4679298 SEND 84.203.148.14:5060 ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 84.203.148.14:5061;rport;branch=z9hG4bKFDC50F318C6B44F09C2FCBBAE4DF34A 9 From: Aisling O' Driscoll sip:[EMAIL PROTECTED]:5061;tag=2391420445 To: sip:[EMAIL PROTECTED];tag=as4ebaa89b Contact: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] CSeq: 6927 ACK Max-Forwards: 70 Content-Length: 0 Original Message From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] xlite and asterisk Date: Wed, 10 Nov 2004 12:39:22 + 404 not found can mean many things, are you using a supporting codec? On Wednesday 10 November 2004 05:25 pm, Ashling O'Driscoll wrote: Hi, Hope somebody can help. I have two xlite clients that register with asterisk. They are called 2000 and 2001. 1)When 2000 rings 2001 a '404 not found' message is returned even though he is registered with asterisk. 2)When 2001 rings 2000, a 'call not approved' error is returned. I found a thread regarding the 'call not approved' error in the asterisk archives but no solution was posted. I have included the relevant portion of my config files below. If any further info is needed please let me know. Also how is it possible to dial a sip address e.g. sip:[EMAIL PROTECTED] from an xlite client? Thanks again, Aisling. sip.conf ;xlite client 1 [2000] type=friend username=2000 secret=whatever nat=yes host=dynamic mailbox=100 [2001] type=friend username=2001 secret=bla nat=yes host=dynamic mailbox=101 extensions.conf exten =3D 2000,1,Dial(SIP/2000,20) exten =3D 2001,1,Dial(SIP/2001,20) ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. - -- - ___ Asterisk-Users mailing list [EMAIL
Re: [Asterisk-Users] Grandstream BugeTone 101 - Multi-Server setup ???
is grandstream still in business?? - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Thursday, November 11, 2004 3:49 AM Subject: [Asterisk-Users] Grandstream BugeTone 101 - Multi-Server setup ??? | | I am reading the manual from Bugetone 101 and found on page 19, the setting | for [8] SIP SP-1 till SP-9 | That would be nice! Could leave the FWD number in place, while I test my | Asterisk setup !! | | However, I did not find out how I can setup SP-2 ~ SP-9 | (Only configured SIP server(s) are displayed) | | I am not at the phone, | | bye | | Ronald | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Aastra/Sayson 480i eval
You can change the dial plan in the .cfg file if you have that on a tftp server. Julian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: 11 November 2004 08:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Aastra/Sayson 480i eval On Wed, 2004-11-10 at 22:25 -0600, Rich Adamson wrote: Just a quick FYI for the Aastra/Sayson 480i SIP phone Just received one and now have it running with *. - Unit came with SIP v1.0.0.34 Release code 0035-00-00 installed. No CDROM shipped with the unit, and a quick look at www.aastra.com and www.sayson.com sites didn't appear as though one can download firmware upgrades. Not sure where one is supposed to get them. There is a little piece of paper that comes with the phone. There it says to contact Sayson support so they can assign you an account on the support site where the firmware is. New firmware is at version 1.0.0.41 There is still a very big problem with this phone, the dial plan will only allow you to dial 10 digits. For local numbers this is not a problem, but you cannot dial long distance. Carlos Chavez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk support for ISDN 1TR6 ?
On Thu, 2004-11-11 at 11:38 +0100, Frank Sautter wrote: can someone give me any hints if the old german ISDN protocol '1TR6' is supported by asterisk. we have a potential customer who has an existing conventional PBX which has to be extended by an asterisk server. unfortunately this existing PBX speaks 1TR6 on it's ISDN ports. I know my Eicon Diva Server BRI card supports 1TR6 on the ISDN side and works fine with Asterisk. To activate 1TR6 all I would have to do is upload the proper firmware to the card. Maybe the AVM Fritz! cards support 1TR6 too. Worth checking out. The Eicon cards are expensive while the AVM Fritz! is much cheaper. Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] high-capacity systems / trouble with Tyan
You will always want a good over-capacity power supply for an AMD server(or any production server for that matter) I always buy nice heavy 500W+ power supplies for all of my servers whether they be AMD or Intel-based. For AMD I've used TTGI power supplies mostly and for Intel I usually use Antec. MATT--- -Original Message- From: Chris A. Icide [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 10, 2004 9:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] high-capacity systems / trouble with Tyan one other question: What kind of power supply do you have in the AMD system? On my 2466, I had alot of problems until I upgraded my power supply to a high quality 500W unit. I seem to remember a while back reading that AMD systems were much more sensitive to power issues that comparable Pentium units. On 02:23 PM 11/10/2004, mattf wrote: Hello, I've had a Tyan dual Athlon MP(2800) machine for a year now and have had several lockups for strange reasons on stock redhat kernel and on custom compiled kernel off of Slackware. I've tried every combination of BIOS settings and changed out all assiciated hardware and found the problem: It's the Tyan. I've also had issues with a couple of SCSI RAID cards when I tried using them with the Tyan card. This all would have really upset me if the Athlon MP platform performed better than the Intel platform, but it doesn't. This Dual Athlon MP system actually handles LESS total Asterisk load than a single P4 3.2 GHz, and the P4 has a lot more Motherboard options and cost much less. This is just my experience, I'm sure I am using Asterisk a little differently than you, I don't have 3 Quad T1 cards in any of my machines, but if that's what you're looking for, I'd suggest the PowerPC(Mac) platform. Asterisk installs just fine right on top of Yellow Dog Linux and the bus speed of a Mac mops the floor with most x86 motherboards, meaning more bandwidth for those bus-hungry Digium boards. MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk support for ISDN 1TR6 ?
well i have an icon diva quadbri card and i already tried uploaded the 1TR6 firmware, which seems to work so far. the problem is, that the capi module and therefore chan_capi do not load correctly. Patrick wrote: I know my Eicon Diva Server BRI card supports 1TR6 on the ISDN side and works fine with Asterisk. To activate 1TR6 all I would have to do is upload the proper firmware to the card. Maybe the AVM Fritz! cards support 1TR6 too. Worth checking out. The Eicon cards are expensive while the AVM Fritz! is much cheaper. On Thu, 2004-11-11 at 11:38 +0100, Frank Sautter wrote: can someone give me any hints if the old german ISDN protocol '1TR6' is supported by asterisk. we have a potential customer who has an existing conventional PBX which has to be extended by an asterisk server. unfortunately this existing PBX speaks 1TR6 on it's ISDN ports. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk DNS issue
Hello all, I just configure Bind 9 in our LAN to resolve the Asterisk name sip.bussines.com for our phones. I want that when a local extensión calls to another local extension, the phone shows Extension@DNS name instead of Extension@ip address like now happens. In all my phones I configure the sip server like sip.bussines.com (dns name), but I don't know how to get it. Someone could give me some hint? any clue will be appreciated. Thanks in advice. Ismael Gil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems in autnenticating with SER / PortaSIP
We have a problem in authenticating with a SIP server running PortaSIP.
first, my exten.conf says:
exten = _396262X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _39064040.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
and sip.conf:
register=390645416983:[EMAIL PROTECTED]/390645416983
[to-uni]
type=peer
secret=XX ; i tried also using md5secret= instead of secret=... but
it's the same
username=390645416983
fromuser=390645416983
host=sip.uni.it
nat=yes
our asterisk pbx correctly registers on sip.uni.it (it is displayed as
registered in sip show registry, and if I issue a sip debug I see the
answer to the registration, correctly reporting the name of the remote
server and our balance:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK1bd57ff9
From: sip:[EMAIL PROTECTED];tag=as4fb9a73e
To: sip:[EMAIL PROTECTED];tag=a4a48d8b20978897d8e0f5c399e6cc29.fbc4
Call-ID: [EMAIL PROTECTED]
CSeq: 103 REGISTER
PortaBilling: available-funds:5.00 currency:EUR
Contact: sip:[EMAIL PROTECTED];q=0.00;expires=115
Server: Sip EXpress router (0.8.14 (i386/freebsd))
Content-Length: 0
The problem is when I try to call a number on the othere side
(39064040): the call is correctly routed, the remote server asks us for
the proper credentials, and it seems to me that asterisk answers their
challenge:
Authorization: Digest username=390645416983, realm=sip.uni.it,
algorithm=MD5, uri=sip:[EMAIL PROTECTED],
nonce=419358e858969bef4a5c77326f2b205b97c672bf,
response=188824ee848f9ed095990999fb2e3893, opaque=
but for some reason it seems that the remote server does not like the
answer. the helpdesk of uni.it says that this is an old bug of asterisk
(actually, the account works with an X-Lite softphone ).
I'm using CVS-v1-0-11/08/04-10:57:05. I hoped that the latest version
corrected this problem as well, but it appears that it is not the case
I enclose the sip debug trace of the call
-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, SIP/[EMAIL PROTECTED]) in
new
stack
We're at 217.18.104.75 port 10880
Answering with preferred capability 0x4(ULAW)
Answering with non-codec capability 0x1(G723)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK2f983a20;rport
From: Roberto Piola sip:[EMAIL PROTECTED];tag=as2fb0ecc6
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 11 Nov 2004 12:14:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 12361 12361 IN IP4 217.18.104.75
s=session
c=IN IP4 217.18.104.75
t=0 0
m=audio 10880 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(NAT) to 217.72.100.4:5060
-- Called [EMAIL PROTECTED]
janis*CLI
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK2f983a20;rport=5060
From: Roberto Piola sip:[EMAIL PROTECTED];tag=as2fb0ecc6
To: sip:[EMAIL PROTECTED];tag=a4a48d8b20978897d8e0f5c399e6cc29.b98a
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
WWW-Authenticate: Digest realm=sip.uni.it,
nonce=419358e858969bef4a5c77326f2b205b97c672bf
Server: Sip EXpress router (0.8.14 (i386/freebsd))
Content-Length: 0
9 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK2f983a20;rport
From: Roberto Piola sip:[EMAIL PROTECTED];tag=as2fb0ecc6
To: sip:[EMAIL PROTECTED];tag=a4a48d8b20978897d8e0f5c399e6cc29.b98a
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to 217.72.100.4:5060
We're at 217.18.104.75 port 10880
Answering with preferred capability 0x4(ULAW)
Answering with non-codec capability 0x1(G723)
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK588b0624;rport
From: Roberto Piola sip:[EMAIL PROTECTED];tag=as2fb0ecc6
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Authorization: Digest username=390645416983, realm=sip.uni.it,
algorithm=MD5, uri=sip:[EMAIL PROTECTED],
nonce=419358e858969bef4a5c77326f2b205b97c672bf,
response=188824ee848f9ed095990999fb2e3893, opaque=
Date: Thu, 11 Nov 2004 12:14:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 12361 12362 IN IP4 217.18.104.75
s=session
c=IN IP4 217.18.104.75
t=0 0
m=audio 10880 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(NAT) to 217.72.100.4:5060
janis*CLI
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK588b0624;rport=5060
From: Roberto Piola sip:[EMAIL PROTECTED];tag=as2fb0ecc6
To: sip:[EMAIL PROTECTED];tag=a4a48d8b20978897d8e0f5c399e6cc29.5919
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
[Asterisk-Users] Several Problems with PhoneJack
Hi! I just bought an ISA phonejack and now I'm having some kind of problems using it. My system: - Debian Woody - P-II/333 - 192mb memory - Kernel 2.6.5 - Asterisk 1.0 (installed as Woody-backport) - slightly modified ixj-module - ISA phonejack - Internet via ADSL (768mBit downlink) I connected asterisk with my sip-account at sipgate.de and connected the phone as well. I can call my phone from the outside (over the sip-account). But my local phone doesn't ring :-( But when I answer the phone - without hearing the ring - the connection is etablished. Outgoing calls (over sip) are initiated aber the first digit. I have read (in the list archieves) that this is a known problem. And there seems to be a patch that with this patch a phone number has to finish with a #. Is this patch available? And my last problem: Outgoing calls seem to have a very bad quality. Is my system too slow? Bye! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk xlite codecs
Hello, I am having problems getting two xlite clients to communicate through asterisk. I am getting an error message: chan_sip.c:2753 process_sdp: No compatible codecs. I have enabled all possible codecs in xlite (Menu - Advanced system settings -Codec settings) and have added the appropriate lines in sip.conf (see below) to allow all codecs. However this is still not working. I have looked this problem up on google and it was previously attributed to old versions of asterisk. However I dont have asterisk setup long and got the most recent version of it. Please help if possible. I must get a call working soon, Kindest Regards, Aisling. sip.conf [general port=5060 bindaddr=0.0.0.0 disallow=all ;xlite client one [2000] type=friend username=2000 secret=bla regexten=2000 nat=yes auth=md5 context=from-sip callerid=Aisling2000 dmtfmode=rfc2833 canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm host=dynamic mailbox=2000 ;xlite client two [2001] type=friend username=2001 secret=bla2 regexten=2001 nat=yes auth=md5 context=from-sip callerid=Julien2001 dmtfmode=rfc2833 canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm host=dynamic mailbox=2001 ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Top posting
Hello, I must admit I live in perpetual fear of forgetting to switch of html or rtf formatting (useful for work) and top posting. I can understand the issue with the former but can see absolutely no reason why top posting is such a problem. In fact I'd far prefer it. I get to my e-mail in batches and bottom posting means I've got to wade through stuff I've just read. I totally agree with snipping extensively. So that I can understand the almost religious fervour on this point could someone please explain to me why top posting is so hated!! Because it's rude to assume that your post must be so important to everyone else that we will all take the time to try to determine the context in which you are making your reply. Top posters force people to page up and down through a message in order to determine the context. And while it may be stuff that *you* have just read, mail does not necessarily arrive in the same order for everyone else, so the replied-to message may not yet have been seen by other participants. Inline quoting allows you to visually skip the quoted material fairly easily (and many participants will do exactly that) *unless* one wants to find out context, which I personally find myself wanting to do maybe a quarter of the time. Top posting makes it impossible to reply to relevant parts within context: you've destroyed the context by doing so. Properly quoted inline text is handled very nicely by good mail clients, colored and highlighted appropriately so it is trivial to see, visually, what is going on. Top posting makes you look like a Microsoft-software-using weenie that is not aware of basic Internet etiquette and who is too lazy to be bothered to conform to basic community standards. Many people, myself included, will simply ditch your message if it becomes too hard to place your message in an appropriate context. I personally follow the spacebar rule at least 95% of the time... within Elm, I use the space bar to progress through mailing list traffic, and that means we only move forward through the text, unless something is /so/ compelling and interesting that it warrants further examination. I can understand that if you are responding to multiple points in an e-mail then you should reply below each point snipping out what is irrelevant to your reply in the original e-mail. If you're responding to an entire e-mail then the proper approach to my mind would to do as you would in business letters and start with a short paragraph explaining what you're doing (e.g. In response to Fred's e-mail about AMD MP motherboards and interrupts, I guess most of us are too lazy to do this so we just leave the original text in the e-mail. If we're really lazy we don't snip the irrelevant stuff out. E-mail is intended to be an easy and informal method for information interchange. We already have a method for providing context, which works without having to summarize someone else's message, and which works through multiple layers of reply (which summarization fails to do concisely). You are *supposed* to be lazy and make use of this more intelligent mechanism, which good software will actually use in order to highlight text based on context, etc. Am I missing something totally?! I'm just about to go and get my flak jacket and helmet in anticipation of the responses. :) Wrap your darn lines at 70. (rat-a-tat-a-tat-a-tat, hope you're wearing that flak jacket! ;-) ) This turns out to be a basic netiquette issue for all the people who have joined the 'net. We did things a little differently in the days of BBS's (though we used in-line message quoting!) but most of us who joined from that community were able to adapt and work with the accepted netiquette of USENET and the Internet. It seems to be mainly the people who joined after the endless september (Google) that have felt that it is more appropriate for the Internet to reshape itself to their own convenience. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk support for ISDN 1TR6 ?
On Thu, 2004-11-11 at 13:18 +0100, Frank Sautter wrote: well i have an icon diva quadbri card and i already tried uploaded the 1TR6 firmware, which seems to work so far. the problem is, that the capi module and therefore chan_capi do not load correctly. [snip] If you could provide some info about your config/setup (asterisk, chan_capi, kernel version etc.) and paste the error perhaps I can help. Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Top posting
Joe Greco schrieb: This turns out to be a basic netiquette issue for all the people who have joined the 'net. Oh yeah ... In every usenet group I'm joining this is an issue because some people don't follow this rule - some other didn't. I personally really hate top posting or ToFu as we say in germany (Text oben Fullquote unten = Text above Quote below). I really don't like to do page up, page down every time to follow a discussion. And I really don't like to play jeopardy, i like to have the questiion _before_ the answers. ;-) We did things a little differently in the days of BBS's (though we used in-line message quoting!) but most of us who joined from that community were able to adapt and work with the accepted netiquette of USENET and the Internet. I really worked. It seems to be mainly the people who joined after the endless september (Google) that have felt that it is more appropriate for the Internet to reshape itself to their own convenience. And there are people who normally do the normal quoting who do top posting with others topposter because they think they can't handle it. I do not like that as well. Bye! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distributed registration SIP/IAX2
Here's a thought, anyone have ideas on how you could take registrations from SIP/IAX users and run an AGI command using Asterisk? My goal would be to enter the user/IP (after user reg's) into a MySQL database then have other asterisk servers read from the same db. This would be for the sake of every server knowing where each user is, a distributed dialplan more a less. As far as I can tell, there's no out-of-box solution for this. If anyone has some code, please share! :-) I heard a rumor that a distributed dialplan was in the works but I can't find any info on this. Thanks, Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: getting callerid from spa3k to asterisk
You could try adding the line insecure=very to the relevant section of the sip.conf this would force asterisk to only validate the IP address and not the user name (possibly but it is woth a shot) Jason On Mon, 8 Nov 2004 10:28:03 -0800, Randy Bush [EMAIL PROTECTED] wrote: You could maybe look at the autocreatepeer option for sip.conf that level of vulnerability would not seem to be a good approach to solving some sort of sip/config problem :-) the problem is in the sip handshake between the spa3k and *. i have been hoping a sip geek would have a chance to look at it. randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Top posting
George Gardiner wrote: I must admit I live in perpetual fear of forgetting to switch of html or rtf formatting (useful for work) and top posting. A: Because otherwise we don't understand what you're replying to. Q: Why top posting is so frowned upon? Cheers, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple NIC's on * box?
Can * support a box with multiple nic cards correctly? Background: small isp operation in the US has a rather large wireless network covering multiple counties. The wireless net is an isolated network using private IP's and nat'ing (via Cisco 7206). Their dsl customers are on another isolated network using registered IP's out to the customer dsl modem (which then does nat'ing) on another Cisco 7206 interface. Will I need to dedicate an * system to each, or can I consider multiple nic's on a single system? (Traffic volumes will be rather low, so multiple machines are not thought to be a requirement now or in the future, unless multiple nic's are not reasonably supported.) Thoughts anyone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Top posting
If someone provides me with an answer to a question or provides information to enhance my asterisk system, I don't care if they top-post or bottom-post. Marv ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Same Extensions in Multiple contexts
On Mon, 8 Nov 2004 20:19:42 -1000, Richard [EMAIL PROTECTED] wrote: I have a question here. If both companies use 200 as their extension, how can * tell which context a received sip call uses? The received sip call will be placed in the context specified buy its definintion in sip.conf Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Top posting
If someone provides me with an answer to a question or provides information to enhance my asterisk system, I don't care if they top-post or bottom-post. That could well be fine, but things rapidly get confusing as it moves from providing a single answer to a simple question to having an extended discussion about some complicated topic, and there are unlimited shades of gray in the middle. It's better to have consistent rules, because inconsistency leads to people who cannot understand the difference between posting a simple answer to a simple question and the 100 screen mega-discussion. There's no reason, other than sheer laziness, to top-post. Providing useful information might lessen the offense somewhat :-), but does not (IMO) make it somehow okay to do. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple NIC's on * box?
It's no issue to use more than one nic. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Thursday, November 11, 2004 7:29 AM To: Asterisk-a-users-list Subject: [Asterisk-Users] Multiple NIC's on * box? Can * support a box with multiple nic cards correctly? Background: small isp operation in the US has a rather large wireless network covering multiple counties. The wireless net is an isolated network using private IP's and nat'ing (via Cisco 7206). Their dsl customers are on another isolated network using registered IP's out to the customer dsl modem (which then does nat'ing) on another Cisco 7206 interface. Will I need to dedicate an * system to each, or can I consider multiple nic's on a single system? (Traffic volumes will be rather low, so multiple machines are not thought to be a requirement now or in the future, unless multiple nic's are not reasonably supported.) Thoughts anyone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] failed to go to next dial command
Hi,
I Looked through tons of pages sofar no luck. Hopefully some one could
tell me the directions or relevant commands for the following.
If I have an outbound call with a normal PSTN number from * to an other
* or IAX provider but that */provider is not reachable because of a
network congestion for example.
Then in that case I would like to go to a next dial command with a small
time out that would use my BRI to push the call out via a regular PSTN
provider:
I guess this is what I mean:
If
exten = _NXX,2,Dial(${IAX2_provider}/${EXTEN})
times out or returns error then dial
exten = _NXX,3,Dial(${CAPI_channel}/${EXTEN}) # goto telco
else
exten = _NXX,4,Congestion/busy/invalid
Is this possible in Asterisk and what should be the approach?
Max.
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RE: [Asterisk-Users] No Inbound CallerID Name Has me Stumped.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Chris Modesitt Sent: Thursday, November 11, 2004 12:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] No Inbound CallerID Name Has me Stumped. My Telco swears that I have Caller ID (Name and Number) being sent to me over our PRI's (I have called them a half dozen times to confirm). My gut feeling is that they are lying to me, this is why. First I decided to Look into my CDR records, they all look like this for incomming calls from the PRI's ,8602144389,8014379394,default,8602144389,Zap/47-1,SIP/8014379394-54ca,Hangup,,2004-11-10 23:35:35,2004-11-10 23:35:56,2004-11-10 23:36:00,25,4,ANSWERED,DOCUMENTATION It appears that I am receiving the CID Number no CID Name however. I have modified my dial plans with a Wait(2) just to make sure the CO has time to send the CallerID before I answer. No luck. Adding the wait will not do anything for that purpose as caller ID name and number is sent on the PRI D channel. Are you sure you have the right protocol set up in your config files? National should send Caller ID name number, if that is what your carrier is using. Try monitoring the D channel to see if names are actually being sent. I thought I saw in an earlier post how to monitor the D channel, but I cant find it right now. If I am missing something or if anybody has any suggestion on how to trouble shoot this further I would greatly appreciate it. Bellow, I have included my zaptel.conf and zapata.conf configuration files:) zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs span=3,3,0,esf,b8zs span=4,4,0,esf,b8zs bchan=1-23,25-47,49-71,73-95 dchan=24,48,72,96 loadzone = us defaultzone=us zapata.conf switchtype=national context=default signalling=pri_cpe group=1 channel = 1-23,25-47,49-71,73-95 Thanks Chris PS I am running CVS Head 08/15/04 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tdm04b outbound call question
Have a newly installed * box (RH ES3, current cvs head) with a tdm04b
(4-port fxo) connected to four US CO Centrex lines. Inbound calls are
being handled correctly via entries shown below. However, outbound
4-digit calling (eg, sip phone dials 8125 or iax2 call dials 8125)
always receives a CO Centrex message ...cannot call this number,
contact your operator I'm 50 miles away from this system and am
trying to debug this remotely. An ordinary analog phone plugged into
the same zap/1 pstn line _can_ dial 8125 and have that centrex
extension rings.
The cli shows:
-- Executing Dial(SIP/6101-0553, Zap/1/8125) in new stack
-- Called 1/8125
-- Zap/1-1 answered SIP/6101-0553
-- Hungup 'Zap/1-1'
Any thoughts on what might be happening here, or how to diagnose the
issue?
zapata.conf
context=inbound-fxo
switchtype=national
signalling=fxs_ks
echocancel=yes
echotraining=800
echocancelwhenbridged=no
usecallerid=no
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
threewaycalling=no
rxgain=0.0
txgain=0.0
callgroup=2
immediate=yes
callprogress=no
musiconhold=default
channel = 1
extensions.conf
[from-sip]
include = outgoing-calls
include = local-extns
include = misc-extns
[outgoing-calls]
exten = _81XX,1,Dial(Zap/1/${EXTEN})
exten = _81XX,102,Dial(Zap/2/${EXTEN})
[inbound-fxo]
exten = s,1,Dial(Sip/6101,15,r)
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Re: [Asterisk-Users] Multiple NIC's on * box?
You can even setup a single nic to have multiple IP addresses in linux... Jesse On Thu, 11 Nov 2004 07:28:30 -0600, Rich Adamson [EMAIL PROTECTED] wrote: Can * support a box with multiple nic cards correctly? Background: small isp operation in the US has a rather large wireless network covering multiple counties. The wireless net is an isolated network using private IP's and nat'ing (via Cisco 7206). Their dsl customers are on another isolated network using registered IP's out to the customer dsl modem (which then does nat'ing) on another Cisco 7206 interface. Will I need to dedicate an * system to each, or can I consider multiple nic's on a single system? (Traffic volumes will be rather low, so multiple machines are not thought to be a requirement now or in the future, unless multiple nic's are not reasonably supported.) Thoughts anyone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk xlite codecs
hi there, How about changing the general conf in sip. disallow=all allow=ulaw allow=alaw allow=gsm not just disallow=all and take them out of the extentions conf. To me since you have the same codecs allowed its kinda not needed in my mind to specify it to that level. Maybe it will fix your problem 2? On Thu, 11 Nov 2004 12:47:07 -, Ashling O'Driscoll [EMAIL PROTECTED] wrote: Hello, I am having problems getting two xlite clients to communicate through asterisk. I am getting an error message: chan_sip.c:2753 process_sdp: No compatible codecs. I have enabled all possible codecs in xlite (Menu - Advanced system settings -Codec settings) and have added the appropriate lines in sip.conf (see below) to allow all codecs. However this is still not working. I have looked this problem up on google and it was previously attributed to old versions of asterisk. However I dont have asterisk setup long and got the most recent version of it. Please help if possible. I must get a call working soon, Kindest Regards, Aisling. sip.conf [general port=5060 bindaddr=0.0.0.0 disallow=all ;xlite client one [2000] type=friend username=2000 secret=bla regexten=2000 nat=yes auth=md5 context=from-sip callerid=Aisling2000 dmtfmode=rfc2833 canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm host=dynamic mailbox=2000 ;xlite client two [2001] type=friend username=2001 secret=bla2 regexten=2001 nat=yes auth=md5 context=from-sip callerid=Julien2001 dmtfmode=rfc2833 canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm host=dynamic mailbox=2001 ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Steven Kalcevich Office +1- 416-576-4457 MSN: [EMAIL PROTECTED] http://www.ciscokid.net http://www.sohonetworks.ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sending SMS from ISDN to cellular
On Wed, Nov 10, 2004 at 11:02:14PM +0100, Elmar Haneke wrote:
how to configure * to send an SMS to an mobile phone (Germany, D2).
In the outgoing directory I do playe an call-file:
Channel: CAPI/[MYMSN]:0106301722270333
http://www.voip-info.org/wiki-Asterisk+cmd+Sms
SMS with T-Com (German Telekom)
Send outgoing messages to 0193010
You have to use the Telekom SMSC as gateway.
Extension: [TARGET-PHONE-NO]
CallerID: Test Test Test
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: smsdial
Priority: 1
You may define some additional variable here and use it as argument
to the sms application:
Callfile SetVar: SmsText=Test Test Test
Extensions.conf SMS(${CALLERIDNUM},,${EXTEN},${SmsText})
New question:
Is it neccessary to register the local ISDN phone number if you want
to receive sms from D2 mobile phones? Short messages from D1
customers are transmitted as sms only if such a registration has
been made.
--
Stefan Tichy [EMAIL PROTECTED]
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RE: [Asterisk-Users] failed to go to next dial command
MvB: Is this possible in Asterisk Yes. and what should be the approach? Read the Wiki ;-) http://www.voip-info.org/wiki-Asterisk+cmd+dial Look at the 'g' parameter. Nick. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra/Sayson 480i eval
- Unit came with SIP v1.0.0.34 Release code 0035-00-00 installed. No CDROM shipped with the unit, and a quick look at www.aastra.com and www.sayson.com sites didn't appear as though one can download firmware upgrades. Not sure where one is supposed to get them. There is a little piece of paper that comes with the phone. There it says to contact Sayson support so they can assign you an account on the support site where the firmware is. New firmware is at version 1.0.0.41 There is still a very big problem with this phone, the dial plan will only allow you to dial 10 digits. For local numbers this is not a problem, but you cannot dial long distance. Thanks. Just tested that and you are absolutely correct. Changing the dialplan to specifically allow a dialed number like 74-9-1-123-4567 fails after the 10th digit (actually starts dialing automatically). Bummer... Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple NIC's on * box?
Cool. I thought that I had seen a few people posting over the last several months that inferred * tied itself to a specific interface, but I must have misread those postings. Thanks. You can even setup a single nic to have multiple IP addresses in linux... Jesse On Thu, 11 Nov 2004 07:28:30 -0600, Rich Adamson [EMAIL PROTECTED] wrote: Can * support a box with multiple nic cards correctly? Background: small isp operation in the US has a rather large wireless network covering multiple counties. The wireless net is an isolated network using private IP's and nat'ing (via Cisco 7206). Their dsl customers are on another isolated network using registered IP's out to the customer dsl modem (which then does nat'ing) on another Cisco 7206 interface. Will I need to dedicate an * system to each, or can I consider multiple nic's on a single system? (Traffic volumes will be rather low, so multiple machines are not thought to be a requirement now or in the future, unless multiple nic's are not reasonably supported.) Thoughts anyone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitor/Record MeetMe Conversations
What is the easiest way to record all parties of a meetme conference into 1 sound file? I tried using Monitor just before the MeetMe call and it gave me files for each person. THanks, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitor/Record MeetMe Conversations
Try to mix them and you will get 1 file ... On Thu, 2004-11-11 at 16:40, Matthew Boehm wrote: What is the easiest way to record all parties of a meetme conference into 1 sound file? I tried using Monitor just before the MeetMe call and it gave me files for each person. THanks, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards Vlad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra/Sayson 480i eval
Just a quick FYI for the Aastra/Sayson 480i SIP phone Just received one and now have it running with *. - Unit came with SIP v1.0.0.34 Release code 0035-00-00 installed. No CDROM shipped with the unit, and a quick look at www.aastra.com and www.sayson.com sites didn't appear as though one can download firmware upgrades. Not sure where one is supposed to get them. www.sayson.com/dealer current firmware is v1.0.0.41 - No apparent support for distinctive ringing or even setting different ring types. options, 3.set ring tone ? - There is an rj11 headphone jack, however to use it one must navigate the screen menu to activate it. (I did not have an rj11 headset to try its use.) There is no front panel button for activating a headset. option, 7.set audio speaker/headset then just use the green speaker button to toggle from handset, speaker, then headset - Appears to be running some sort of Linux kernel. VxWorks from Wind River it also has a a telnet interface where you can access every option help console dir nvram showparm to read/show cfg readfile cfg name.cfg set seq # from showparm value ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco poe
I know this is on the wiki, I just want to confirm so I don't blow up my cisco phones. I've got several cisco 7940's all running using cisco power cubes. However, my boss wants me to switch just a few over to poe, but doesn't want to fork out the dough for a nice cisco poe switch, or anybody else's poe switch for that matter. So my question is, what is the '99.999% sure/safe' poe injector solution that most people are using for the cisco phones? Right now I'm looking at buying the 3-Com 3CNJPSE (qty 2-3) to power the few specific locations where a power cube just wouldn't look right, like a conference room table for example. :) I know this solution, thanks to the fact that it is a 'hack', is far from the 99.999% I just stated, but it also seems to be the only low-end solution for poe. Am I right, or just plain blind? Thanks, Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administratordba Sparco.com Email: [EMAIL PROTECTED] 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053 Fax: (901) 872 8482 USA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No SIP registration but user has dialled out?!?
There is an autocreatepeer flag in the sip.conf http://voip-info.org/wiki-Asterisk+sip+autocreatepeer That allows calls to go through without having to register. Race Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: 11 November 2004 03:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] No SIP registration but user has dialled out?!? hi when looking into the sipfriends table (using mysql sipfriends from asterisk cvs version -r v1-0), I see timestamp and ipaddr set to 0/NULL. When looking into the CDR, the user has dialled out recently. Also sip show peer xxx shows no data. How can this be true? roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra/Sayson 480i eval
There is still a very big problem with this phone, the dial plan will only allow you to dial 10 digits. For local numbers this is not a problem, but you cannot dial long distance. edit the dial plan in the cfg file # The dial plan that the 480i phone should use # Where, # 0, 1, 2, 3, 4, 5, 6, 7, 8, 9: is a Digit symbol # 'x': matches any digit symbol (wildcard), # '+': matches zero or more of the preceding digit symbol or # [] expression # []: Symbol inclusive OR # '-': used only with [], represent a range of acceptable symbols # '*', '#': match the keypad symbols. sip dial plan: or telnet xxx.xxx.xxx.xxx admin/pwd console showparm see the sequence number for sip dial plan set 172 911|011x|1011xx|9[1-9]xx|1[2-9]x|7[1-9] xx|xx+*|xx+#|*xx|#xx+# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cisco poe
Christopher http://www.voip-info.org/tiki-print.php?page=Cisco+POE Its ALWAYS on the wiki :) Good question, but the 7940 is NOT a proper 802.3af (POE) device. It is a polarity problem, which can be fixed with a crimp tool. With 1 minute of crimping I have seen them work with the DLINK injectors. Cheers Cian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher L. Wade Sent: Thursday, November 11, 2004 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] cisco poe I know this is on the wiki, I just want to confirm so I don't blow up my cisco phones. I've got several cisco 7940's all running using cisco power cubes. However, my boss wants me to switch just a few over to poe, but doesn't want to fork out the dough for a nice cisco poe switch, or anybody else's poe switch for that matter. So my question is, what is the '99.999% sure/safe' poe injector solution that most people are using for the cisco phones? Right now I'm looking at buying the 3-Com 3CNJPSE (qty 2-3) to power the few specific locations where a power cube just wouldn't look right, like a conference room table for example. :) I know this solution, thanks to the fact that it is a 'hack', is far from the 99.999% I just stated, but it also seems to be the only low-end solution for poe. Am I right, or just plain blind? Thanks, Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administratordba Sparco.com Email: [EMAIL PROTECTED] 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053 Fax: (901) 872 8482 USA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco poe
I know this is on the wiki, I just want to confirm so I don't blow up my cisco phones. I've got several cisco 7940's all running using cisco power cubes. However, my boss wants me to switch just a few over to poe, but doesn't want to fork out the dough for a nice cisco poe switch, or anybody else's poe switch for that matter. So my question is, what is the '99.999% sure/safe' poe injector solution that most people are using for the cisco phones? Right now I'm looking at buying the 3-Com 3CNJPSE (qty 2-3) to power the few specific locations where a power cube just wouldn't look right, like a conference room table for example. :) I know this solution, thanks to the fact that it is a 'hack', is far from the 99.999% I just stated, but it also seems to be the only low-end solution for poe. Am I right, or just plain blind? The thing you need to watch out for is that the 7940/7960 do not do PoE, or (more specifically) do not do standard PoE. They use the Cisco variant, a pre-standard PoE which has reversed polarity. This means that, unless a switch actually claims to be compatible with the Cisco variant, a non-Cisco PoE switch will not power the 7940/7960. Likewise, injectors probably won't work, though some people report success with various hacks such as wiring up a cable to provide a Cisco-compatible layout. Beware that there are severe risks in doing this, in that if you inadvertently plug in some other PoE device to such a port, or don't clearly mark such cables, that you are likely to burn up some other PoE device at a future point in time. Bad Cisco, very bad Cisco... ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple NIC's on * box?
Yes, Look in the wiki for bindaddr bindaddr = 0.0.0.0 :IP Address to bind to (listen on) http://voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.con f Be careful with the bind address. I know I have been burned by not getting it right. Asterisk answers on eth0 but I am routing to eth1, the calls won't go and the registeration won't work. It will drive you crazy. Best Idea. Draw a map with all the address on it on a piece of paper. Race Vanderdecken * ate Vanderdecken DOT combine -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: 11 November 2004 08:29 To: Asterisk-a-users-list Subject: [Asterisk-Users] Multiple NIC's on * box? Can * support a box with multiple nic cards correctly? Background: small isp operation in the US has a rather large wireless network covering multiple counties. The wireless net is an isolated network using private IP's and nat'ing (via Cisco 7206). Their dsl customers are on another isolated network using registered IP's out to the customer dsl modem (which then does nat'ing) on another Cisco 7206 interface. Will I need to dedicate an * system to each, or can I consider multiple nic's on a single system? (Traffic volumes will be rather low, so multiple machines are not thought to be a requirement now or in the future, unless multiple nic's are not reasonably supported.) Thoughts anyone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco poe
By far the best poe (price/performance) I have seen for Cisco poe (or standard poe) is the Netgear FSM7326P. http://www.cdw.com/shop/products/default.aspx?EDC=568864 It is a managed layer3 poe switch (24 port) with 2 gigabit ports also. Works out of the box with Cisco and Snoms (it auto detects which polarity they want). No adapters needed for either. And it is about $1100. We are using 4 of them and love them. Jeb Campbell [EMAIL PROTECTED] Christopher L. Wade wrote: I know this is on the wiki, I just want to confirm so I don't blow up my cisco phones. I've got several cisco 7940's all running using cisco power cubes. However, my boss wants me to switch just a few over to poe, but doesn't want to fork out the dough for a nice cisco poe switch, or anybody else's poe switch for that matter. So my question is, what is the '99.999% sure/safe' poe injector solution that most people are using for the cisco phones? Right now I'm looking at buying the 3-Com 3CNJPSE (qty 2-3) to power the few specific locations where a power cube just wouldn't look right, like a conference room table for example. :) I know this solution, thanks to the fact that it is a 'hack', is far from the 99.999% I just stated, but it also seems to be the only low-end solution for poe. Am I right, or just plain blind? Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Monitor/Record MeetMe Conversations
What is the easiest way to record all parties of a meetme conference into 1
sound file?
The easiest way is to Originate a call from the manager interface from a
Local extension that is setup to record(see example below) for a flat amount
of time and have it call into the meetme room. It'll record all sides of the
conversation. We've been using this for months and we do over 1000
recordings a day like this.
Recording Extension:
# send the callerID string in the originate to name the recording file
exten = 8309,1,Answer
exten = 8309,2,Monitor(wav,${CALLERIDNAME})
exten = 8309,3,Wait,3600
exten = 8309,4,Hangup
MATT---
I tried using Monitor just before the MeetMe call and it gave me files for
each person.
THanks,
Matthew
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Re: [Asterisk-Users] Top posting
Joe Greco wrote: *snipped There's no reason, other than sheer laziness, to top-post. Providing useful information might lessen the offense somewhat :-), but does not (IMO) make it somehow okay to do. ... JG so if by chance there is a thread you are interested in that 3 other TP'er were engaged in, you would BP, when you KNEW they would all 'continue' the thread TP'n. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Special Characters In Passwords
Hello, I have a brief question, how do you format the following line in the sip.conf file, the # in it seems to throw it off, but I have no option but to keep it on the password register = 1999555:[EMAIL PROTECTED] I tried escaping the #, but I still can't get it to work Thanks Doug Eubanks [EMAIL PROTECTED] *** DISCLAIMER *** This e-mail and any attachments thereto may contain information, which is confidential and/or protected by intellectual property rights and are intended for the sole use of the recipient(s) named above. Any use of the information contained herein (including, but not limited to, total or partial reproduction, communication or distribution in any form) by persons other than the designated recipient(s) is prohibited. If you have received this e-mail in error, please notify the sender either by telephone or by e-mail and delete the material from any computer. Thank you for your cooperation. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Top posting
Joe Greco wrote: There's no reason, other than sheer laziness, to top-post. Providing useful information might lessen the offense somewhat :-), but does not (IMO) make it somehow okay to do. so if by chance there is a thread you are interested in that 3 other TP'er were engaged in, you would BP, BP? Bottom-post, maybe? No. Bottom posting is nearly as stupid. We have these neat standards for doing inline quoting. Like here, where I've broken in mid-sentence, adding a quote character to the quoted line below, so I can be very precise about who said what. when you KNEW they would all 'continue' the thread TP'n. As long as I'm not in a rush, I typically *fix* formatting that I disapprove of. :-) You will notice some reformatting of your quoted reply, including the removal of some gratuitous lines... The goal should be able to make it easy to follow the thread of discussion. When the material being replied to is more than maybe a dozen or so lines away, it becomes more difficult to follow the flow. That's a very loose rule of thumb, of course, since there are many examples where someone may post dozens of lines of text, and then have an equally large reply, but the general idea remains. Regards, ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hooking up a an Adit 600
Thank you very much for your response. I was wondering if it would be ok for me to ask you a couple of additional questions. 1. Do you think this woul work? http://www.phonegeeks.com/patpanwit25p.html 2. If I use the 25 pair (Amphenol) for hooking up analog phones, what ports on the ADIT 600 do I use for hooking up my eight analog incoming phone lines? Thanks again for your help. If my questions are unclear (not suprising since I am completely clueless) feel free to call me toll free at 888-448-7874. Richard Reina. --- Brent Franks [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Richard Reina Sent: Wednesday, November 10, 2004 3:27 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Hooking up a an Adit 600 I have purchased an Adit 600 but with 6 FXS 8 channel cards. Can somone tell me where I plug analog phones in. The cards do not have any ports on them. You can get an amphenol 25 pair cable and connect it to a punchdown block that also has an amphenol connector. From there you can then punch down phone jacks. - B ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] xlite and asterisk
It's been awhile since I've played with X-Lite, but I think it absolutely *has* to use the MD5 auth stuff. Use md5secret rather than secret in sip.conf. You'll have to MD5 hash your password... there's documentation on this in the Wiki. -Chad On Nov 10, 2004, at 9:25 AM, Ashling O'Driscoll wrote: Hi, Hope somebody can help. I have two xlite clients that register with asterisk. They are called 2000 and 2001. 1)When 2000 rings 2001 a '404 not found' message is returned even though he is registered with asterisk. 2)When 2001 rings 2000, a 'call not approved' error is returned. I found a thread regarding the 'call not approved' error in the asterisk archives but no solution was posted. I have included the relevant portion of my config files below. If any further info is needed please let me know. Also how is it possible to dial a sip address e.g. sip:[EMAIL PROTECTED] from an xlite client? Thanks again, Aisling. sip.conf ;xlite client 1 [2000] type=friend username=2000 secret=whatever nat=yes host=dynamic mailbox=100 [2001] type=friend username=2001 secret=bla nat=yes host=dynamic mailbox=101 extensions.conf exten =3D 2000,1,Dial(SIP/2000,20) exten =3D 2001,1,Dial(SIP/2001,20) ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. --- - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Top posting
[snip] It's somewhat amusing, but mostly annoying, to see people fighting this fight still even after 10+ years on the Internet. In my experience, there will always be 2 kinds of posters in email lists/USENET: 1) The somewhat intelligent comprehensive types who understand inline posting and the reasoning behind it. 2) The lazy type who don't give a damn and top post because its easier. These are the types who usually have that annoying feature in Windows turned on that automatically zaps the mouse cursor to whatever default button appears in modal dialog boxes. The sad thing is that the people in group #1 never seem to realize that no matter how many people in group #2 they might convert in their crusades, there will always be more group 2'ers in line behind them, and in the meantime they piss off everyone else in group #1. -- Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] setup of cisco 7960 phone tftp asking for unkown file
Found the setup docs to convert cisco to SIP phone. setup tftp downloaded version 7.3 from cisco, put in /tftpboot directory. reset the phone. looked at the /var/log/messages and found this: Nov 11 16:35:21 snorkel in.tftpd[4465]: RRQ from 192.168.1.85 filename OS79XX.TXT Nov 11 16:35:21 snorkel in.tftpd[4466]: RRQ from 192.168.1.85 filename SEP000FF78DEBB2.cnf.xml [EMAIL PROTECTED] tftpboot]# I dont know what the format is for the SEP-MACADDRESS.cnf.xml file is Anybody? Thanks, Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] working Marconi sys X config
OK, the line's now set to ETSI, still having probs. Anyone got some working configs ? Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] setup of cisco 7960 phone tftp asking for unkown file
According to: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip3 The phone should request the OS79XX.txt file from the TFTP server, and after that should download the new firmware, and it shouldn't request the SEPcnf.xml file. Are you sure that the OS79XX.txt file is in place correctly? I think it's the file responsible for telling the phone that a new firmware file is available. -Chris On Thu, 11 Nov 2004 11:42:30 -0500, Jerry Geis [EMAIL PROTECTED] wrote: Found the setup docs to convert cisco to SIP phone. setup tftp downloaded version 7.3 from cisco, put in /tftpboot directory. reset the phone. looked at the /var/log/messages and found this: Nov 11 16:35:21 snorkel in.tftpd[4465]: RRQ from 192.168.1.85 filename OS79XX.TXT Nov 11 16:35:21 snorkel in.tftpd[4466]: RRQ from 192.168.1.85 filename SEP000FF78DEBB2.cnf.xml [EMAIL PROTECTED] tftpboot]# I dont know what the format is for the SEP-MACADDRESS.cnf.xml file is Anybody? Thanks, Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring Asterisk As A Sip Server
Yes, you can do this, in fact I'm sure most of the people who use asterisk do this. Check out http://www.voip-info.org/tiki-index.php?page=Asterisk for more information about how to set up SIP channels and users. -Chris On Tue, 09 Nov 2004 00:19:54 +0500, Adnan Ahmed [EMAIL PROTECTED] wrote: Hello Group, I want to configure my Asterisk Server As a SIP is there any possibality.How i do that.Any help is highly appreciated. Thanks in advance. Regards Adnan . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] setup of cisco 7960 phone tftp asking for unkownfile
Here are the files in the directory. [EMAIL PROTECTED] tftpboot]# ls cisco.P0S3-07-3-00.zip OS79XX.TXT P003-07-3-00.bin P003-07-3-00.sbn P0S3-07-3-00.loads P0S3-07-3-00.sb2 SEP000FF78DEBB2.cnf [EMAIL PROTECTED] tftpboot]# According to: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip3 The phone should request the OS79XX.txt file from the TFTP server, and after that should download the new firmware, and it shouldn't request the SEPcnf.xml file. Are you sure that the OS79XX.txt file is in place correctly? I think it's the file responsible for telling the phone that a new firmware file is available. -Chris On Thu, 11 Nov 2004 11:42:30 -0500, Jerry Geis geisj at pagestation.com wrote: Found the setup docs to convert cisco to SIP phone. setup tftp downloaded version 7.3 from cisco, put in /tftpboot directory. reset the phone. looked at the /var/log/messages and found this: Nov 11 16:35:21 snorkel in.tftpd[4465]: RRQ from 192.168.1.85 filename OS79XX.TXT Nov 11 16:35:21 snorkel in.tftpd[4466]: RRQ from 192.168.1.85 filename SEP000FF78DEBB2.cnf.xml [root at snorkel tftpboot]# I dont know what the format is for the SEP-MACADDRESS.cnf.xml file is Anybody? Thanks, Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] setup of cisco 7960 phone tftp asking for unkownfile
I just tried the tftp localhost and "get OS79XX.TXT" it says access violation. Here are the permissions of the files. any idea on why I'm getting access violation? drw-r--r-- 2 nobody nobody 4096 Nov 11 11:35 tftpboot [EMAIL PROTECTED] tftpboot]# [EMAIL PROTECTED] tftpboot]# ls -l /tftpboot/ total 1436 -rw-r--r-- 1 nobody nobody 582861 Nov 11 09:55 cisco.P0S3-07-3-00.zip -rw-r--r-- 1 nobody nobody 15 Nov 2 15:47 OS79XX.TXT -rw-r--r-- 1 nobody nobody 129416 Nov 2 15:47 P003-07-3-00.bin -rw-r--r-- 1 nobody nobody 129820 Nov 2 15:47 P003-07-3-00.sbn -rw-r--r-- 1 nobody nobody 459 Nov 2 15:41 P0S3-07-3-00.loads -rw-r--r-- 1 nobody nobody 592414 Nov 2 15:55 P0S3-07-3-00.sb2 -rw-r--r-- 1 nobody nobody 3195 Nov 11 11:35 SEP000FF78DEBB2.cnf [EMAIL PROTECTED] tftpboot]# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DISA() context restrictions
On Thu, 11 Nov 2004 09:33:29 +0200 (SAST), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Tue, 9 Nov 2004, Michael George wrote: The only difference to my extensions.conf file is that if I have: exten = s,2,DISA(no-password, disa) -- Executing DISA(IAX2/[EMAIL PROTECTED]/6, no-password| disa) in new stack Nov 9 19:50:33 DEBUG[14521]: app_disa.c:160 disa_exec: Context: disa Bet you its the space after the comma. Notice that the Context: disa has two spaces. So try DISA(no-password,disa) without the space and see if that helps. If it does, its obviously a bug, but you have a work-around at least. Steve I wouldn't really call that a bug, especially since I've seen cautions in several places against including spaces. It's just the way it is, one wouldn't include spaces in a CSV file, nor inbetween comma seperated values in the GECOS field in /etc/passwd, so why between arguments in the dial plan. No fault of Michael George of course, he didn't know that was the case before but now he does... I just wouldn't call it a bug. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Preventing Call Forwarding by SIP UA
[Apologies if this is a repost, I needed to subscribe to post through GMANE.] I have a use case where I must not allow/respect or at least restrict the SIP 302 Moved Temporarily message that many SIP UAs send when the user enables Call Forwarding. This is because some calls are personal to the user and some a system related or coming to a group and should not be, for example, forwarded to a mobile phone. I need this on a per-call basis. Anyone have any ideas? I've asked around and Googled to no avail. Thanks, A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-OH323 OUTCODEC
Try:
SetGlobalVar(OH323_OUTCODEC=g723.1)
Michael.
M. Ehsanul Karim wrote:
Hello,
What would be the outcodec value for g723.1 (6.3k). I have g723
support which works with SIP (not pass thru) , but when I use OH323 it
always
Unsupported ${OH323_OUTCODEC} value (G72316K3)!
I have enabled all g723 in oh323.conf
SetGlobalVar(OH323_OUTCODEC=G72316K3)
Regards,
Ehsan
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Re: [Asterisk-Users] Multiple NIC's on * box?
I've had problems using bind to bind to only my lan interface on eth1. It has no problem when I specify 0.0.0.0 it binds to all. On Thu, 11 Nov 2004 10:36:55 -0500, Race Vanderdecken [EMAIL PROTECTED] wrote: Yes, Look in the wiki for bindaddr bindaddr = 0.0.0.0 :IP Address to bind to (listen on) http://voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.con f Be careful with the bind address. I know I have been burned by not getting it right. Asterisk answers on eth0 but I am routing to eth1, the calls won't go and the registeration won't work. It will drive you crazy. Best Idea. Draw a map with all the address on it on a piece of paper. Race Vanderdecken * ate Vanderdecken DOT combine -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: 11 November 2004 08:29 To: Asterisk-a-users-list Subject: [Asterisk-Users] Multiple NIC's on * box? Can * support a box with multiple nic cards correctly? Background: small isp operation in the US has a rather large wireless network covering multiple counties. The wireless net is an isolated network using private IP's and nat'ing (via Cisco 7206). Their dsl customers are on another isolated network using registered IP's out to the customer dsl modem (which then does nat'ing) on another Cisco 7206 interface. Will I need to dedicate an * system to each, or can I consider multiple nic's on a single system? (Traffic volumes will be rather low, so multiple machines are not thought to be a requirement now or in the future, unless multiple nic's are not reasonably supported.) Thoughts anyone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No Inbound CallerID Name Has me Stumped.
On Wed, Nov 10, 2004 at 11:48:58PM -0700, Chris Modesitt wrote: My Telco swears that I have Caller ID (Name and Number) being sent to me over our PRI's (I have called them a half dozen times to confirm). My gut feeling is that they are lying to me, this is why. First I decided to Look into my CDR records, they all look like this for incomming calls from the PRI's ,8602144389,8014379394,default,8602144389,Zap/47-1,SIP/8014379394-54ca,Hangup,,2004-11-10 23:35:35,2004-11-10 23:35:56,2004-11-10 23:36:00,25,4,ANSWERED,DOCUMENTATION It appears that I am receiving the CID Number no CID Name however. I have modified my dial plans with a Wait(2) just to make sure the CO has time to send the CallerID before I answer. No luck. If I am missing something or if anybody has any suggestion on how to trouble shoot this further I would greatly appreciate it. Try checking out libpri-matt from CVS and see if you get CID name. I've been working on CID name over facility message implemenation and it's possible that they are using facility IEs in the SETUP message of the call that has the CID name info. If it still doesn't work tell, give me a `pri debug span x` (where x is the span number that the call comes in on) and I might be able to help you. Matthew Fredrickson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astGUIclient Problem -- http://10.10.10.15/astguiclient/admin.php
Hello, I was trying to install astGUIclient following the SCRATCH INSTALLATION document. After I finished Step (6.1) -- creating the MySQL asterisk database and try to do http://10.10.10.15/astguiclient/admin.php, it failed. The following are the warning or error messages: Any idea where is the problem? My work-around solution is to hard-code the Asterisk IP Address in the /usr/local/apache2/htdocs/astguiclient/dbconnect.php file. Thanks Ken == Warning: mysql_connect(): Access denied for user: '[EMAIL PROTECTED]' (Using password: YES) in /usr/local/apache2/htdocs/astguiclient/dbconnect.php on line 3 Warning: mysql_query(): supplied argument is not a valid MySQL-Link resource in /usr/local/apache2/htdocs/astguiclient/admin.php on line 41 Warning: mysql_fetch_row(): supplied argument is not a valid MySQL result resource in /usr/local/apache2/htdocs/astguiclient/admin.php on line 42 Warning: Cannot modify header information - headers already sent by (output started at /usr/local/apache2/htdocs/astguiclient/dbconnect.php:3) in /usr/local/apache2/htdocs/astguiclient/admin.php on line 52 Warning: Cannot modify header information - headers already sent by (output started at /usr/local/apache2/htdocs/astguiclient/dbconnect.php:3) in /usr/local/apache2/htdocs/astguiclient/admin.php on line 53 Invalid Username/Password: ||| |SELECT count(*) from phones where login='' and pass='' and active = 'Y' and status='ADMIN';| === -- ___ Find what you are looking for with the Lycos Yellow Pages http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.asp?SRC=lycos10 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Multiple NIC's on * box?
Rich Adamson wrote: Cool. I thought that I had seen a few people posting over the last several months that inferred * tied itself to a specific interface, but I must have misread those postings. Thanks. I have a bunch of Asterisk systems using VLANs to reach multiple subnets over a single physical NIC. Works very well! A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Callerid is recieved by fxo, but sometimes not passed to extensions
Are you waiting until the start of the second ring cycle before
answering the phone?
CLID information is sent in-band between the first and second ring
cycles. If you interrupt this process (by answering the phone before
transmission is complete), you will not receive the CLID information.
[EMAIL PROTECTED] wrote:
Hi,
I'm having a problem with callerid. It is recieved fine by
the fxo (it appears in the cdr, and voicemail app gets it
fine), but it is passed to the internal phones works about 25% of the
time. The internal phones are all analog, a dvg-1120M (mgcp
firmware) and a quicknet phonejack. There seems to be no
pattern, as the same number will sometimes appear or not.
I've tried forcing the callerid by placing a
1,SetCallerID(${CALLERID}) before dialing the channel. This did not
help.
Any suggestions?
Thanks,
-Ry
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[Asterisk-Users] Re: No SIP registration but user has dialled out?!?
Race Vanderdecken wrote: when looking into the sipfriends table (using mysql sipfriends from asterisk cvs version -r v1-0), I see timestamp and ipaddr set to 0/NULL. When looking into the CDR, the user has dialled out recently. Also sip show peer xxx shows no data. How can this be true? A registration is not required to place calls, only the correct authentication. Registration is required to *receive* calls. A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] astGUIclient Problem -- http://10.10.10.15/a stguiclient/admin.php
I was trying to install astGUIclient following the SCRATCH INSTALLATION document. After I finished Step (6.1) -- creating the MySQL asterisk database and try to do http://10.10.10.15/astguiclient/admin.php, it failed. The following are the warning or error messages: Any idea where is the problem? You might just need to run this statement in mysql: GRANT SELECT,INSERT,UPDATE,DELETE on asterisk.* to [EMAIL PROTECTED] IDENTIFIED BY 1234; For some reason MySQL doesn't always see localhost as fitting the % wildcard and you have to explicitly give localhost users permissions. MATT--- Thanks Ken == Warning: mysql_connect(): Access denied for user: '[EMAIL PROTECTED]' (Using password: YES) in /usr/local/apache2/htdocs/astguiclient/dbconnect.php on line 3 Warning: mysql_query(): supplied argument is not a valid MySQL-Link resource in /usr/local/apache2/htdocs/astguiclient/admin.php on line 41 Warning: mysql_fetch_row(): supplied argument is not a valid MySQL result resource in /usr/local/apache2/htdocs/astguiclient/admin.php on line 42 Warning: Cannot modify header information - headers already sent by (output started at /usr/local/apache2/htdocs/astguiclient/dbconnect.php:3) in /usr/local/apache2/htdocs/astguiclient/admin.php on line 52 Warning: Cannot modify header information - headers already sent by (output started at /usr/local/apache2/htdocs/astguiclient/dbconnect.php:3) in /usr/local/apache2/htdocs/astguiclient/admin.php on line 53 Invalid Username/Password: ||| |SELECT count(*) from phones where login='' and pass='' and active = 'Y' and status='ADMIN';| === -- ___ Find what you are looking for with the Lycos Yellow Pages http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.asp ?SRC=lycos10 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_capi patch : fax support
Hi everybody,
Anybody could give me a little hint to apply the patch described below and
how to enable sfftobmp ? reading the post below, fax.php seems to be used to
mail the result but was not able to find it, do I have to write it ?
Thanks in advance,
jl
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Carl Sempla
Envoyé : jeudi 4 novembre 2004 17:31
À : [EMAIL PROTECTED]
Objet : [Asterisk-Users] chan_capi patch : fax support
Hello,
For those of you who have a CAPI card with an on-board DSP (like some Eicon
Diva Server), this patch allows you to receive faxes.
If you want to answer a channel in fax mode, use capiAnswerFax() instead of
Answer()
If you use Answer(), you will be in voice mode. If the hardware DSP detects
a fax tone, you can switch from voice to fax mode by calling
capiAnswerFax().
Example of use :
line number 123, play something, if a fax tone is detected, handle it
line number 124, answer directly in fax mode
[incoming]
exten = 123,1,Answer()
exten = 123,2,BackGround(jpop)
exten = 124,1,Goto(handle_fax,s,1)
exten = fax,1,Goto(handle_fax,s,1)
[handle_fax]
exten = s,1,capiAnswerFax(/tmp/${UNIQUEID})
exten = s,2,Hangup()
exten = h,1,deadagi,fax.php // Run sfftobmp and mail it.
The output of capiAnswerFax is a SFF file. Use sfftobmp to convert it.
With a Diva Server, theses features are allowed : fax up to 33600, high
resolution. Color Fax /JPEG Compression is disabled (I can't test it).
You can download the patch at :
http://www.mlkj.net/asterisk/chan_capi-0.3.5-patch.tar.bz2
A fix for a dead lock issue is also included (Oct 22 18:06:00
WARNING[11275]: channel.c:472 ast_channel_walk_locked: Avoided initial
deadlock for 'CAPI[contr1/173720007]/7', 10 retries!)
--
Carl
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Re: [Asterisk-Users] Zaptel module load errors under stock Fedora Core 2 (2.6.8-1.521 kernel )
Just a reminder, if you are using the stock fedora kernel I'd recommend rebuilding it without preemption turned off as I've experience kernel panics from the zaptel driver. Digium tech support agrees (or at least did a few weeks ago) that is was a problem. Adam Sean Kennedy wrote: Got it, that was it. Thank you so much Adam. For those searching, here's the solution: vi /usr/src/linux-2.6/Makefile Remove the word 'custom' from the version information. If you've been following along at home, you'll need to `make clean` in the kernel source directory. Then, `make prepare-all`. Granted, `make clean` probably isn't really required, but I do it too often to avoid problems. Then, go `make clean` in the zap directory, and `make linux26`, `make install`, and we're gold. Thanks again for your help Adam. Sean Sean Kennedy wrote: Thank you, Adam. I think I see how to do that ( the kernel Makefile has that version information. So either I just change that and recompile zap, or I have to recompile the kernel AND zap. As long as it works, I'm happy ). Question: I can force the zaptel module to load, but I can't force the wcfxo module. Would this indicate that it's not finding the hardware card, or would this module load regardless of the hardware in the machine? Thank you again. Sean Adam Fineberg wrote: This appears to be a module version mismatch. Notice that the kernel is linux-2.6.8-1.521 but the modules are 2.6.8-1.521custom. This means you need to remake your modules or your kernel to get them to match. Also, you should try rebuilding the kernel with preemption turned off. It helps avoid a zaptel crash. Adam Sean Kennedy wrote: Hi folks, start to finish, this is what I did: cd /usr/src/linux-2.6.8-1.521 make prepare-all cd .. wget http://www.asterisk.org/zaptel-1.0.0.tar.gz tar xfsz zaptel-1.0.0.tar.gz cd zaptel-1.0.0 less README less README.Linux26 ( see, I really did RTFM ;) ) ln -s /usr/src/linux-2.6.8-1.521 /usr/src/linux-2.6 mv /lib/modules/`uname -r`/build /lib/modules/`uname -r`/build.bak ( There was a preexisting build directory ) ln -s /usr/src/linux-2.6.8-1.521 /lib/modules/`uname -r`/build make linux26 make install modprobe wcfxo And this is what I get when I try to load the modules: WARNING: Error inserting zaptel (/lib/modules/2.6.8-1.521/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.8-1.521/misc/zaptel.ko): Invalid module format FATAL: Error inserting wcfxo (/lib/modules/2.6.8-1.521/misc/wcfxo.ko): Invalid module format FATAL: Error running install command for wcfxo And this shows up in my /var/log/messages: Nov 10 17:25:35 firewall kernel: zaptel: version magic '2.6.8-1.521custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.8-1.521 686 REGPARM 4KSTACKS gcc-3.3' Nov 10 17:26:11 firewall kernel: zaptel: version magic '2.6.8-1.521custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.8-1.521 686 REGPARM 4KSTACKS gcc-3.3' Nov 10 17:26:11 firewall kernel: zaptel: version magic '2.6.8-1.521custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.8-1.521 686 REGPARM 4KSTACKS gcc-3.3' Nov 10 17:26:11 firewall kernel: wcfxo: version magic '2.6.8-1.521custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.8-1.521 686 REGPARM 4KSTACKS gcc-3.3' Me being me, and this being a test machine, I tried `modprobe -f wcfxo`, and this is what I got: FATAL: Error inserting wcfxo (/lib/modules/2.6.8-1.521/misc/wcfxo.ko): Invalid module format FATAL: Error running install command for wcfxo Now, as to what I am trying to do: I have a generic intel 537 card that I was hoping to use as a generic fxo(?). It works on Suse 9.1, but I am running into problems on this fc2 box. I imagine if I can just get the zaptel module to load without any brute force, I'd be ok. Any help that can be offered I greatly apprecaite. Sean Kennedy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] astGUIclient Problem --http://10.10.10.15/astguiclient/admin.php
The same happened to me on an old RH9 It´s a permission stuff.. Check mysql permissions for root, cron.. Also check passwords (you can connect to mysql without password). You can edit dbconnect.php to use another user (ex: root) Guido Rebert Network Manager GrupoPyD - +54 11 4800 -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Ken Chan Enviado el: Jueves, 11 de Noviembre de 2004 02:29 p.m. Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] astGUIclient Problem --http://10.10.10.15/astguiclient/admin.php Hello, I was trying to install astGUIclient following the SCRATCH INSTALLATION document. After I finished Step (6.1) -- creating the MySQL asterisk database and try to do http://10.10.10.15/astguiclient/admin.php, it failed. The following are the warning or error messages: Any idea where is the problem? My work-around solution is to hard-code the Asterisk IP Address in the /usr/local/apache2/htdocs/astguiclient/dbconnect.php file. Thanks Ken == Warning: mysql_connect(): Access denied for user: '[EMAIL PROTECTED]' (Using password: YES) in /usr/local/apache2/htdocs/astguiclient/dbconnect.php on line 3 Warning: mysql_query(): supplied argument is not a valid MySQL-Link resource in /usr/local/apache2/htdocs/astguiclient/admin.php on line 41 Warning: mysql_fetch_row(): supplied argument is not a valid MySQL result resource in /usr/local/apache2/htdocs/astguiclient/admin.php on line 42 Warning: Cannot modify header information - headers already sent by (output started at /usr/local/apache2/htdocs/astguiclient/dbconnect.php:3) in /usr/local/apache2/htdocs/astguiclient/admin.php on line 52 Warning: Cannot modify header information - headers already sent by (output started at /usr/local/apache2/htdocs/astguiclient/dbconnect.php:3) in /usr/local/apache2/htdocs/astguiclient/admin.php on line 53 Invalid Username/Password: ||| |SELECT count(*) from phones where login='' and pass='' and active = 'Y' and status='ADMIN';| === -- ___ Find what you are looking for with the Lycos Yellow Pages http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.asp ?SRC=lycos10 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.788 / Virus Database: 533 - Release Date: 01/11/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.788 / Virus Database: 533 - Release Date: 01/11/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 190/220 dialplan strings?
Anyone have an example dialplan string as to what is valid for these phones. Their admin manual doesn't cover it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Palm Tungsten and Asterisk
Hello, Maybe someone here can help me. I am looking for VoIP software ( client ) on my Palm Tungsten. So I can make use of my Palm and Asterisk server. Thank you for help. Bartosz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem using Digi DataFire Micro V
Hi all, i'm experiencing a problem using a Digi DataFire Micro V ISDN card. I can't dial out nor recieve a call. *CLI dial Nov 11 18:18:13 NOTICE[-151061888]: channel.c:284 ast_alloc_uniqueid: uid = asterisk-2806-1100193493.0 -- Executing Dial(OSS/dsp, Zap/g1/||trT) in new stack Nov 11 18:18:13 NOTICE[-186840144]: channel.c:284 ast_alloc_uniqueid: uid = asterisk-2806-1100193493.1 -- Called g1/0925545119 -- Channel 0/1, span 1 got hangup -- Hungup 'Zap/1-1' == No one is available to answer at this time -- Timeout on OSS/dsp == CDR updated on OSS/dsp -- Executing Goto(OSS/dsp, #|1) in new stack -- Goto (local,#,1) -- Executing Playback(OSS/dsp, demo-thanks) in new stack Console call has been answered -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(OSS/dsp, ) in new stack == Spawn extension (local, #, 2) exited non-zero on 'OSS/dsp' Hangup on console *CLI Nov 11 18:19:31 WARNING[-174232656]: chan_zap.c:7275 zt_pri_error: PRI: !! Got a UA, but i'm in state 0 Nov 11 18:19:31 WARNING[-174232656]: chan_zap.c:7275 zt_pri_error: PRI: !! Got S-frame while link down Segmentation fault ___- SO: Fedora Core 2 with kernel 2.6.8-521 __ zaptel.conf # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3 __ zapata.conf [channels] language=it switchtype = euroisdn signalling = bri_net_ptmp pridialplan=local prilocaldialplan=local pritrustusercid = yes echocancel=yes immediate=yes group = 1 context=default channel = 1-2 in /var/log/messages i find Nov 11 17:41:35 pbx kernel: Zapata Telephony Interface Registered on major 196 Nov 11 17:41:35 pbx zaptel: Loading zaptel framework: succeeded Nov 11 17:41:36 pbx kernel: ACPI: PCI interrupt :00:07.0[A] - GSI 10 (level, low) - IRQ 10 Nov 11 17:41:36 pbx kernel: zaphfc: Digi International Digi DataFire Micro V (Europe) configured at mem 0x22914f00 fifo 0xac78000(0x8c78000) IRQ 10 HZ 1000 Nov 11 17:41:36 pbx kernel: zaphfc: Card 0 configured for NT mode Nov 11 17:41:36 pbx kernel: zaphfc: 1 hfc-pci card(s) in this box. Nov 11 17:41:36 pbx kernel: Registered tone zone 11 (Italy) Nov 11 17:41:36 pbx ztcfg: Nov 11 17:41:36 pbx ztcfg: Zaptel Configuration Nov 11 17:41:36 pbx ztcfg: == Nov 11 17:41:36 pbx ztcfg: Nov 11 17:41:36 pbx ztcfg: SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Nov 11 17:41:36 pbx ztcfg: Nov 11 17:41:36 pbx ztcfg: 3 channels configured. Nov 11 17:41:36 pbx ztcfg: Nov 11 17:41:36 pbx zaptel: Running ztcfg: succeeded Nov 11 17:44:40 pbx kernel: application asterisk uses obsolete OSS audio interface Nov 11 17:55:26 pbx kernel: zaphfc: empty HDLC frame received. Nov 11 17:56:07 pbx last message repeated 120 times Nov 11 18:00:52 pbx kernel: zaphfc: stop Nov 11 18:00:52 pbx kernel: zaphfc: shutting down card at 22914f00. Nov 11 18:00:52 pbx kernel: Debug: sleeping function called from invalid context at include/linux/rwsem.h:66 Nov 11 18:00:52 pbx kernel: in_atomic():0[expected: 0], irqs_disabled():1 Nov 11 18:00:52 pbx kernel: [0211b765] __might_sleep+0x82/0x8c Nov 11 18:00:52 pbx kernel: [0222de0f] class_device_del+0x20/0xb0 Nov 11 18:00:52 pbx kernel: [0222dea7] class_device_unregister+0x8/0x10 Nov 11 18:00:52 pbx kernel: [22fb491b] zt_unregister+0x9b/0x1b0 [zaptel] Nov 11 18:00:52 pbx kernel: [021de330] pci_disable_device+0x20/0x4f Nov 11 18:00:52 pbx kernel: [22df7178] hfc_shutdownCard+0x178/0x1f0 [zaphfc] Nov 11 18:00:52 pbx kernel: [0211f1aa] printk+0x277/0x2ed Nov 11 18:00:52 pbx kernel: [22df8be6] cleanup_module+0x116/0x1a0 [zaphfc] Nov 11 18:00:52 pbx kernel: [021366a6] try_stop_module+0x16/0x1b Nov 11 18:00:52 pbx kernel: [02136845] sys_delete_module+0x129/0x170 Nov 11 18:00:52 pbx kernel: [02151db2] unmap_vma_list+0xe/0x17 Nov 11 18:00:52 pbx kernel: [0215218a] do_munmap+0x1d8/0x1e2 Nov 11 18:00:52 pbx kernel: [021181a7] do_page_fault+0x0/0x489 Nov 11 18:00:52 pbx kernel: unregistered from zaptel. Nov 11 18:00:52 pbx kernel: zaphfc: freed one card. Nov 11 18:00:52 pbx kernel: Zapata Telephony Interface Unloaded Nov 11 18:00:52 pbx zaptel: Removing zaptel module: succeeded __ Any suggestion is welcome. Thank's in advance Regards Accursio Avona ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Palm Tungsten and Asterisk
XTEN http://www.xten.com the same people that make x-lite make a softphone for handhelds. I use it on my handheld with pocket pc 2003. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Thursday, November 11, 2004 12:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Palm Tungsten and Asterisk Hello, Maybe someone here can help me. I am looking for VoIP software ( client ) on my Palm Tungsten. So I can make use of my Palm and Asterisk server. Thank you for help. Bartosz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Palm Tungsten and Asterisk
XTEN http://www.xten.com the same people that make x-lite make a softphone for handhelds. I use it on my handheld with pocket pc 2003. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Thursday, November 11, 2004 12:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Palm Tungsten and Asterisk Hello, Maybe someone here can help me. I am looking for VoIP software ( client ) on my Palm Tungsten. So I can make use of my Palm and Asterisk server. I have it on my PocketPC. But it does not work on PALM OS. I need something like that on Palm. Bartosz Thank you for help. Bartosz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: NoOp
What is the purpose of NoOp (no operation) if it does nothing?
among other things, it logs, so you can see a context being
entered. e.g.
[ext-foo]
exten = _X.,1,NoOp(ext-foo cid=${CALLERIDNUM})
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Re: [Asterisk-Users] Distributed registration SIP/IAX2
Matt Schulte wrote: Here's a thought, anyone have ideas on how you could take registrations from SIP/IAX users and run an AGI command using Asterisk? My goal would be to enter the user/IP (after user reg's) into a MySQL database then have other asterisk servers read from the same db. This would be for the sake of every server knowing where each user is, a distributed dialplan more a less. As far as I can tell, there's no out-of-box solution for this. If anyone has some code, please share! :-) I heard a rumor that a distributed dialplan was in the works but I can't find any info on this. Have a look at DUNDi. There is an entry on the wiki (www.voip-info.org) and I think www.dundi.com -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iconnect incoming problems
Steave, OK, so they made changes to register string. I never had user number in my register string. It was always; register=1408215:[EMAIL PROTECTED] It worked that way for about 11 months. anyway when I included the user number, it started sending me invite messages again. Thnkyou for this great advice. Cheers Sathya -Original Message- From: Steve Rubin [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 10, 2004 6:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] iconnect incoming problems Sathya Weerasooriya wrote: Hi, I cannot receive any calls via icoonect. I can make outgoing calls, and also I can see sipauth.deltathree.com registering me correctly (I am on public internet). When I try calling-in I wouldn't even get an invite my way. I then hookup a grandstream ata and without a problem it was able to receive calls. I have been using Iconnect for months without many problems. I was using asterisk 1.0 when I detected the problem and just upgraded to 1.0.11 (latest stable) but still the problem persists. What does your register look like? Mine is... register=1408215::[EMAIL PROTECTED] = Phone Number = Password = User Number ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] setup of cisco 7960 phone tftp asking for unkown
Found the setup docs to convert cisco to SIP phone. setup tftp downloaded version 7.3 from cisco, put in /tftpboot directory. reset the phone. looked at the /var/log/messages and found this: Nov 11 16:35:21 snorkel in.tftpd[4465]: RRQ from 192.168.1.85 filename OS79XX.TXT Nov 11 16:35:21 snorkel in.tftpd[4466]: RRQ from 192.168.1.85 filename SEP000FF78DEBB2.cnf.xml [EMAIL PROTECTED] tftpboot]# I dont know what the format is for the SEP-MACADDRESS.cnf.xml file is Anybody? UTWL (Use The Wiki, Luke) http://www.voip-info.org/wiki-Firmware+issues+on+7940+-+7960 ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Palm Tungsten and Asterisk
Try here.. http://www.vliusa.com/prof_personal/index.php -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Thursday, November 11, 2004 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Palm Tungsten and Asterisk XTEN http://www.xten.com the same people that make x-lite make a softphone for handhelds. I use it on my handheld with pocket pc 2003. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Thursday, November 11, 2004 12:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Palm Tungsten and Asterisk Hello, Maybe someone here can help me. I am looking for VoIP software ( client ) on my Palm Tungsten. So I can make use of my Palm and Asterisk server. I have it on my PocketPC. But it does not work on PALM OS. I need something like that on Palm. Bartosz Thank you for help. Bartosz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel module load errors under stock FedoraCore 2 (2.6.8-1.521 kernel )
Please clarify: Fedore Core - build with preemption off or preemption on ? The way you worded it, it's almost as if you're suggesting it with it turned on? Thanks! Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Fineberg Sent: Thursday, November 11, 2004 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaptel module load errors under stock FedoraCore 2 (2.6.8-1.521 kernel ) Just a reminder, if you are using the stock fedora kernel I'd recommend rebuilding it without preemption turned off as I've experience kernel panics from the zaptel driver. Digium tech support agrees (or at least did a few weeks ago) that is was a problem. Adam Sean Kennedy wrote: Got it, that was it. Thank you so much Adam. For those searching, here's the solution: vi /usr/src/linux-2.6/Makefile Remove the word 'custom' from the version information. If you've been following along at home, you'll need to `make clean` in the kernel source directory. Then, `make prepare-all`. Granted, `make clean` probably isn't really required, but I do it too often to avoid problems. Then, go `make clean` in the zap directory, and `make linux26`, `make install`, and we're gold. Thanks again for your help Adam. Sean Sean Kennedy wrote: Thank you, Adam. I think I see how to do that ( the kernel Makefile has that version information. So either I just change that and recompile zap, or I have to recompile the kernel AND zap. As long as it works, I'm happy ). Question: I can force the zaptel module to load, but I can't force the wcfxo module. Would this indicate that it's not finding the hardware card, or would this module load regardless of the hardware in the machine? Thank you again. Sean Adam Fineberg wrote: This appears to be a module version mismatch. Notice that the kernel is linux-2.6.8-1.521 but the modules are 2.6.8-1.521custom. This means you need to remake your modules or your kernel to get them to match. Also, you should try rebuilding the kernel with preemption turned off. It helps avoid a zaptel crash. Adam Sean Kennedy wrote: Hi folks, start to finish, this is what I did: cd /usr/src/linux-2.6.8-1.521 make prepare-all cd .. wget http://www.asterisk.org/zaptel-1.0.0.tar.gz tar xfsz zaptel-1.0.0.tar.gz cd zaptel-1.0.0 less README less README.Linux26 ( see, I really did RTFM ;) ) ln -s /usr/src/linux-2.6.8-1.521 /usr/src/linux-2.6 mv /lib/modules/`uname -r`/build /lib/modules/`uname -r`/build.bak ( There was a preexisting build directory ) ln -s /usr/src/linux-2.6.8-1.521 /lib/modules/`uname -r`/build make linux26 make install modprobe wcfxo And this is what I get when I try to load the modules: WARNING: Error inserting zaptel (/lib/modules/2.6.8-1.521/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.8-1.521/misc/zaptel.ko): Invalid module format FATAL: Error inserting wcfxo (/lib/modules/2.6.8-1.521/misc/wcfxo.ko): Invalid module format FATAL: Error running install command for wcfxo And this shows up in my /var/log/messages: Nov 10 17:25:35 firewall kernel: zaptel: version magic '2.6.8-1.521custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.8-1.521 686 REGPARM 4KSTACKS gcc-3.3' Nov 10 17:26:11 firewall kernel: zaptel: version magic '2.6.8-1.521custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.8-1.521 686 REGPARM 4KSTACKS gcc-3.3' Nov 10 17:26:11 firewall kernel: zaptel: version magic '2.6.8-1.521custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.8-1.521 686 REGPARM 4KSTACKS gcc-3.3' Nov 10 17:26:11 firewall kernel: wcfxo: version magic '2.6.8-1.521custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.8-1.521 686 REGPARM 4KSTACKS gcc-3.3' Me being me, and this being a test machine, I tried `modprobe -f wcfxo`, and this is what I got: FATAL: Error inserting wcfxo (/lib/modules/2.6.8-1.521/misc/wcfxo.ko): Invalid module format FATAL: Error running install command for wcfxo Now, as to what I am trying to do: I have a generic intel 537 card that I was hoping to use as a generic fxo(?). It works on Suse 9.1, but I am running into problems on this fc2 box. I imagine if I can just get the zaptel module to load without any brute force, I'd be ok. Any help that can be offered I greatly apprecaite. Sean Kennedy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED]
[Asterisk-Users] broadvoice patch and 16 second re-registers
Hi, With the Patch, now I see following log notices every 13-14 seconds on my console for each SIP provider. Nov 10 22:52:06 NOTICE[1089948224]: chan_sip.c:4023 sip_reregister:-- Re-registration for [EMAIL PROTECTED] Nov 10 22:52:06 NOTICE[1089948224]: chan_sip.c:6795 handle_response: Outbound Registration: Expiry for sip.broadvoice.com is 159 sec (Scheduling reregistration in 144000 ms) Nov 10 22:52:19 NOTICE[1089948224]: chan_sip.c:4023 sip_reregister:-- Re-registration for [EMAIL PROTECTED] Is this desirable, that asterisk now have to send reregister every 13 seconds. I thought it was previousely in minutes. Besides now we have this annoying Notice. If this patch is needed for NAT users as Steave Sokol mentioned, then there should be an option when this fix get to CVS so that it only applies to Broadvoice Context when * behind NAT. Sathya ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] xlite and asterisk
X-Lite works fine for me with plain text passwords. Unlike the stuff below, though, I'm not using nat=yes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Scott Sent: Thursday, November 11, 2004 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] xlite and asterisk It's been awhile since I've played with X-Lite, but I think it absolutely *has* to use the MD5 auth stuff. Use md5secret rather than secret in sip.conf. You'll have to MD5 hash your password... there's documentation on this in the Wiki. -Chad On Nov 10, 2004, at 9:25 AM, Ashling O'Driscoll wrote: Hi, Hope somebody can help. I have two xlite clients that register with asterisk. They are called 2000 and 2001. 1)When 2000 rings 2001 a '404 not found' message is returned even though he is registered with asterisk. 2)When 2001 rings 2000, a 'call not approved' error is returned. I found a thread regarding the 'call not approved' error in the asterisk archives but no solution was posted. I have included the relevant portion of my config files below. If any further info is needed please let me know. Also how is it possible to dial a sip address e.g. sip:[EMAIL PROTECTED] from an xlite client? Thanks again, Aisling. sip.conf ;xlite client 1 [2000] type=friend username=2000 secret=whatever nat=yes host=dynamic mailbox=100 [2001] type=friend username=2001 secret=bla nat=yes host=dynamic mailbox=101 extensions.conf exten =3D 2000,1,Dial(SIP/2000,20) exten =3D 2001,1,Dial(SIP/2001,20) ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. -- - - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk DNS issue
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ismaelg Sent: Thursday, November 11, 2004 6:46 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk DNS issue Hello all, I just configure Bind 9 in our LAN to resolve the Asterisk name sip.bussines.com for our phones. I want that when a local extensión calls to another local extension, the phone shows Extension@DNS name instead of Extension@ip address like now happens. In all my phones I configure the sip server like sip.bussines.com (dns name), but I don't know how to get it. Someone could give me some hint? any clue will be appreciated. I think that is going to completely depend on the phone or softphone you are using. Did you create a PTR record as well to do the reverse lookup? That might help as well. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Inbound CallerID Name Has me Stumped.
Thanks Matt, I will give that a shot tonight and will let you knowJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Testing H323
Hello Can or is there somewhere a way to test my outgoing H323 I like to connect to a terminating server but I'm still getting hangups. Phone is ringing on the othersite but my asterisk telling my no one availble at this moment. Like to test my H323 loutgoing line. I't looks so stuppid if something wrong on my site. Using Nufone H323 compiled pwlib and openH323 correct.. My error messsages. -- Executing Dial(SIP/4786042-aa0e, H323/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- H323/69.XX.XX.XX is ringing == No one is available to answer at this time -- H323/69.XX.XX.XX answered SIP/142-aa0e Nov 11 19:50:54 NOTICE[1144222912]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible == Spawn extension (terminator, , 1) exited non-zero on 'SIP/142-aa0e' Thanks -- Dit bericht is gescand op virussen en gevaarlijke content en is veilig bevonden. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialplan question - doesn't quite work
Hi - I have a zaptel card with 4 modules - 2 fxs and two fxo. I have two phone lines coming into my house. For now I want an incoming call to ring a phone here, and then if no answer to ring another number (by calling out on the other line) for 15 seconds... then if no answer send to voicemail. It seems to work, except the last part... the outgoing call doesn't time out... if not answered it will ring for eternity. exten = s,1,Answer exten = s,2,playback(thx4call) exten = s,3,Dial(Zap/1|15,t) ; Calls channel 1 exten = s,4,playback(trying_bert) exten = s,5,Dial(Zap/4/12168810880|15,r) exten = s,6,Voicemail,u100 exten = s,7,hangup exten = s,104,Voicemail,b100 exten = s,105,hangup exten = s,106,Voicemail,u100 exten =? s,107,hangup I am a newbie but have done lots of reading and playing around. ANy advice is welcome. DB ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Inbound CallerID Name Has me Stumped.
Matt, I am unable to check-out libpri-matt, is there something special I need to do? Let me know and Thanks! cvs server: cannot find module `libpri-matt' - ignored cvs [checkout aborted]: cannot expand modules From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Modesitt Sent: Thursday, November 11, 2004 12:12 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] No Inbound CallerID Name Has me Stumped. Thanks Matt, I will give that a shot tonight and will let you knowJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZT_CHANCONFIG failed on channel 1: No such device or address (6)
I bought a Wildcard TDM400P earlier this week. I compiled the software from CVS and installed it. When ztcfg runs I get the error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) After checking /proc/pci I don't see the board. Why wouldn't the board be showing up? Its in a slightly older machine (with an i440BX chipset) that any other PCI card I have works fine in. My brand new desktop sees it. Any ideas? Thanks, Rob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel module load errors under stock Fedora Core 2 (2.6.8-1.521 kernel )
Well, from what I'm looking at here, it appears preemption is off by default ( installed the sources, did make menuconfig. *shrug* Thanks again Sean Adam Fineberg wrote: Just a reminder, if you are using the stock fedora kernel I'd recommend rebuilding it without preemption turned off as I've experience kernel panics from the zaptel driver. Digium tech support agrees (or at least did a few weeks ago) that is was a problem. Adam Sean Kennedy wrote: Got it, that was it. Thank you so much Adam. For those searching, here's the solution: vi /usr/src/linux-2.6/Makefile Remove the word 'custom' from the version information. If you've been following along at home, you'll need to `make clean` in the kernel source directory. Then, `make prepare-all`. Granted, `make clean` probably isn't really required, but I do it too often to avoid problems. Then, go `make clean` in the zap directory, and `make linux26`, `make install`, and we're gold. Thanks again for your help Adam. Sean Sean Kennedy wrote: Thank you, Adam. I think I see how to do that ( the kernel Makefile has that version information. So either I just change that and recompile zap, or I have to recompile the kernel AND zap. As long as it works, I'm happy ). Question: I can force the zaptel module to load, but I can't force the wcfxo module. Would this indicate that it's not finding the hardware card, or would this module load regardless of the hardware in the machine? Thank you again. Sean Adam Fineberg wrote: This appears to be a module version mismatch. Notice that the kernel is linux-2.6.8-1.521 but the modules are 2.6.8-1.521custom. This means you need to remake your modules or your kernel to get them to match. Also, you should try rebuilding the kernel with preemption turned off. It helps avoid a zaptel crash. Adam Sean Kennedy wrote: Hi folks, start to finish, this is what I did: cd /usr/src/linux-2.6.8-1.521 make prepare-all cd .. wget http://www.asterisk.org/zaptel-1.0.0.tar.gz tar xfsz zaptel-1.0.0.tar.gz cd zaptel-1.0.0 less README less README.Linux26 ( see, I really did RTFM ;) ) ln -s /usr/src/linux-2.6.8-1.521 /usr/src/linux-2.6 mv /lib/modules/`uname -r`/build /lib/modules/`uname -r`/build.bak ( There was a preexisting build directory ) ln -s /usr/src/linux-2.6.8-1.521 /lib/modules/`uname -r`/build make linux26 make install modprobe wcfxo And this is what I get when I try to load the modules: WARNING: Error inserting zaptel (/lib/modules/2.6.8-1.521/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.8-1.521/misc/zaptel.ko): Invalid module format FATAL: Error inserting wcfxo (/lib/modules/2.6.8-1.521/misc/wcfxo.ko): Invalid module format FATAL: Error running install command for wcfxo And this shows up in my /var/log/messages: Nov 10 17:25:35 firewall kernel: zaptel: version magic '2.6.8-1.521custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.8-1.521 686 REGPARM 4KSTACKS gcc-3.3' Nov 10 17:26:11 firewall kernel: zaptel: version magic '2.6.8-1.521custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.8-1.521 686 REGPARM 4KSTACKS gcc-3.3' Nov 10 17:26:11 firewall kernel: zaptel: version magic '2.6.8-1.521custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.8-1.521 686 REGPARM 4KSTACKS gcc-3.3' Nov 10 17:26:11 firewall kernel: wcfxo: version magic '2.6.8-1.521custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.8-1.521 686 REGPARM 4KSTACKS gcc-3.3' Me being me, and this being a test machine, I tried `modprobe -f wcfxo`, and this is what I got: FATAL: Error inserting wcfxo (/lib/modules/2.6.8-1.521/misc/wcfxo.ko): Invalid module format FATAL: Error running install command for wcfxo Now, as to what I am trying to do: I have a generic intel 537 card that I was hoping to use as a generic fxo(?). It works on Suse 9.1, but I am running into problems on this fc2 box. I imagine if I can just get the zaptel module to load without any brute force, I'd be ok. Any help that can be offered I greatly apprecaite. Sean Kennedy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
