Re: [asterisk-users] You won't help me anymore?
ABBAS SHAKEEL wrote: why don't you post your question On Sun, Jan 10, 2010 at 4:42 PM, hadi motamedi motamed...@gmail.com mailto:motamed...@gmail.com wrote: On Sun, Jan 10, 2010 at 10:58 AM, Gergo Csibra csi...@gmail.com mailto:csi...@gmail.com wrote: Sunday, January 10, 2010, 11:24:22 AM, hadi wrote: You are not willing to help me anymore ? Why do you think this? -- Best regards, Gergomailto:csi...@gmail.com mailto:csi...@gmail.com Thank you for your reply . I am facing with callerId problem on my sip inbound calls , so I strongly need your technical help . Can you please help me ? Yes, post your question clear and consicely, include all relevant information and snip all unneccessary history. Note that: no reply != not wanting to help... It *is* obviously possible people just do not KNOW the answer!... (Oh what shock and horror!!!) -- Francesco -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
Rick Green wrote: On Thu, 7 Jan 2010, David Gibbons wrote: Yes, gmail DOES default to top posting, because bottom posting is silly (in general, but especially for a client that hides quoted text (like gmail)). Top posting is modern. And better. And doesn't make me scroll through 10 thousand messages and awful rsa keys to get to the message... FLAME AWAY!!! This is not intended as a flame... I just got a gmail account a month ago, and haven't used it but for a single google group and calendar notifications. This morning, after seeing the above message, I actually hit reply on several messages, and this is what I found: 1) In every case, gmail presented me with the entire text of the message in the compose window. There was NO indication of 'hidden' full-quote. Yes, the cursor is initially placed at the top of the window. 2) The 'Daily Agenda' mails I get from Google Calendar arrive in some kind of rich formatting, but right at the top of the composer window is a small unobtrusive link labelled 'Plain text', which strips the formatting, and makes deleting the unnecessary text trivial. 3) Plain text email arriving from a friend's android/gmail device are displayed in plain text already. 4) I searched thru the settings dialog, and I found nothing where I had explicitly told it to include the text in a reply, or to show or hide that text. I DID specify that 'plain text' was to be my default outgoing format. IMHO, top-posting isn't the problem, but just an obvious symptom of the real problem, which is failure to edit/strip the quotes to the bare minimum. When a thread gets hijacked by top-posters, who bang out their thoughts without even scrolling down to see all the garbage below, another problem also becomes apparent, and that is the failure of many MUAs to honor 'sigdashes', which is the convention of preceeding your sigfile with a line that is 'dash dash space CR'. A compliant MUA will strip that line and everything after it when quoting for a reply or forward. Note for the list admin: Please preceed your message-footer with a sigdashes line! And to add on to this: aside from whether you think it is silly or not, there are: 1) RFC's 2) List rules And when both of those tell you to bottom-post, then who are you to decide otherwise, just because you think it is silly? Well, maybe I think it is silly that I cannot hit you in the face everytime you say I, would you allow me to hit you, or would you protest and demand I keep to the rules that tell me I can't do that? Civility demands I keep to the rules and do not hit you in the face. The same civility demands you keep to the rules as well and do not top-post! Is that *really* so hard? Just because Microsoft and others decide to place the cursor at the wrong position doesn't mean you have to be a mindless herd-animal and follow that incorrect behavior! Please people, stop these totally pointless discussions and get back on-topic!... PS: I did not have to cut anything, thanks to Rick using the dash-dash-space convention, and Thunderbird honoring that convention. PPS: Top or Bottom posting does NOT change anything about the fact you should SNIP stuff that is no longer relevant Just my €0.02! -- Francesco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
Steve Totaro wrote: read your posting and it will tell you haw to remove yourself. On Thu, Jan 7, 2010 at 10:49 AM, Rick Dean ric.d...@gmail.com mailto:ric.d...@gmail.com wrote: Can I be taken off the mailing list please. Thanks. rick http://lists.digium.com/mailman/listinfo/asterisk-users And a proper mail client will also parse the headers and provide unsubscribe information/buttons based on that... --FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
Dan Journo wrote: I've never seen that in Outlook. What client do you use? Lately I have been using Thunderbird with an RFC2369 header plugin. --FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk recieves 11 when pressing 1 from SIPphone
jonas kellens wrote: [Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid extension '11', but no rule 'i' in context ...[snip]... When testing IVR and pressing 1 from my Grandstream SIP-phone, the above message is printed on the Asterisk CLI. How come Asterisk receives my 1 as 11 ?? Settings in my SIP-phone are : Send DTFM : via RTP(rfc2833) via SIP INFO [Dec 31 10:45:40] WARNING[17928]: pbx.c:2518 __ast_pbx_run: Invalid extension '33', but no rule 'i' in context ...[snip]... Same problem when pressing 3... Thank you. Jonas. It may be me, but it looks like Asterisk correctly interprets the information, as the phone is configured to send both via RTP (once) and SIP INFO (twice). Your config tells the phone to send the digits twice, so Asterisk sees them twice... 1 twice makes 11, 3 twice makes 33! Try changing the phone's config to only use either RTP *or* SIP INFO... Good luck! --FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iphone client app
Alex Samad wrote: On Tue, Dec 15, 2009 at 08:59:34PM +0100, Benny Amorsen wrote: Gavin Spurgeon gspurg...@dageek.co.uk writes: iSip (£2.39) http://itunes.apple.com/gb/app/isip-push-service-formerly-sipphone/id298202722?mt=8 I have been very impressed by the audio quality from iSip, at least from the other end so to speak. It shares the basic flaw of not being able to run in the background with every other iPhone app. They try to can't you use backgrounder ? He probably could, but that is assuming he's jailbroken his phone... Not everybody sees a need to do that, though backgrounder by itself would be a very good reason to do it... Best, --FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automon = *1 one touch recording
Joseph wrote: On 12/08/09 11:11, Jared Smith wrote: On Mon, 2009-12-07 at 19:39 -0700, Joseph wrote: After pressing *1 console is not showing anything indicating that the call is being recorded: I find that I often have to adjust the featuredigittimeout setting in features.conf, as users tend to take their time between the * and 1 keys when turning on automon. -- Jared Smith Digium, Inc. Well, ;transferdigittimeout = 3 (default is 3 seconds) but this does not work or does not take any effect, this feature worked perfectly in Asterisk 1.2 I just tried it, I set: transferdigittimeout = 4 it doesn't work. I'm using cordless phone and I'm 100% sure that it take me less then 1.5 seconds to press *1 with one finger. However, when I tried pressing *1 using two fingers it worked. So, it seems to me transferdigittimeout setting doesn't work or doesn't take any effect. Hmmm... That would possibly also explain why I always succeed in doing *2 xfers, and my wife always fails... I always have 2 fingers on those buttons, and she is the single-finger-typing-kind'o'gal... Weird though that unattended (##) xfers DO work for her as well... --FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] my kernel is dazed and confused
Dr. Michael J. Chudobiak wrote: Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue Would my Digium TDM410P cause an NMI, or is my computer failing? - Mike Googling for the error seems to indicate a possible kernel bug... Are you using Ubuntu or Debian?... -- Francesco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voipbuster not ringing, other SIP peers are ringing...
Francesco Peeters wrote: Francesco Peeters wrote: Does anybody else see the same behavior for VoipBuster connections? When I trace one of the other SIP peers, I see it sends this message: -- --- SIP read from 82.101.62.99:5060 --- SIP/2.0 180 Ringing Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl Contact: sip:82.101.62.99:5060 Content-Type: application/sdp CSeq: 103 INVITE From: "**" sip:***...@sip.xs4all.nl;tag=as70e84199 Record-Route: sip:82.101.62.115;lr;r2=on;ftag=as70e84199,sip:82.101.63.5;lr;r2=on;ftag=as70e84199 Server: Cirpack/v4.41b (gw_sip) To: sip:0031*...@sip.xs4all.nl;tag=00-08168-044b6f36-245cd72c7 Via: SIP/2.0/UDP ***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92 Content-Length: 182 v=0 o=cp10 125193221174 125193221174 IN IP4 82.101.62.66 s=SIP Call c=IN IP4 194.109.8.2 t=0 0 m=audio 36984 RTP/AVP 8 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=ptime:20 a=sendrecv - --- (12 headers 10 lines) --- Found RTP audio format 8 Peer audio RTP is at port 194.109.8.2:36984 Found audio description format PCMA for ID 8 Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.109.8.2:36984 -- SIP/*-089ca9b8 is ringing -- SIP/*-089ca9b8 is making progress passing it to IAX2/2104-2287 Scheduling destruction of SIP dialog '740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 82.101.62.99:5060: CANCEL sip:0031**...@sip.xs4all.nl SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport From: "**" sip:**...@sip.xs4all.nl;tag=as70e84199 To: sip:0031**...@sip.xs4all.nl Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- However when I dial exactly the same from VoipBuster, I see this instead: -- --- SIP read from 77.72.169.129:5060 --- SIP/2.0 183 Session progress Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport From: "*" sip:**...@sip.voipbuster.com;tag=as1374705a To: sip:0031**...@sip.voipbuster.com;tag=120113ac4a54a269af9e2c Contact: sip:0031**...@77.72.169.129:5060 Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 162 v=0 o=* 1251932194 1251932194 IN IP4 194.221.62.33 s=SIP Call c=IN IP4 194.221.62.33 t=0 0 m=audio 8958 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 - --- (11 headers 8 lines) --- Found RTP audio format 0 Peer audio RTP is at port 194.221.62.33:8958 Found audio description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.221.62.33:8958 -- SIP/-089dc538 is making progress passing it to IAX2/2104-8077 == Connect attempt from '127.0.0.1' unable to authenticate Scheduling destruction of SIP dialog '1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 77.72.169.129:5060: CANCEL sip:0031**...@sip.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport From: "**" sip:***...@sip.voipbuster.com;tag=as1374705a To: sip:0031**...@sip.voipbuster.com Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- As you can see, there are different packets being sent, and in the 2nd case, there is no "is ringing" message, which is rather irritating... Any suggestions would be appreciated... TIA BTW: I am talking about the ringtone the caller should hear... The other side is ringing, and calls are established just fine, but it is very irritating to hear nothing until the call either fails or gets picked up... NM! Found out this only happens on a single extension, and that one was using IAX... Changed it to SIP as well and got ringing there too! -- FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register No
[asterisk-users] Voipbuster not ringing, other SIP peers are ringing...
Does anybody else see the same behavior for VoipBuster connections? When I trace one of the other SIP peers, I see it sends this message: -- --- SIP read from 82.101.62.99:5060 --- SIP/2.0 180 Ringing Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl Contact: sip:82.101.62.99:5060 Content-Type: application/sdp CSeq: 103 INVITE From: ** sip:***...@sip.xs4all.nl;tag=as70e84199 Record-Route: sip:82.101.62.115;lr;r2=on;ftag=as70e84199,sip:82.101.63.5;lr;r2=on;ftag=as70e84199 Server: Cirpack/v4.41b (gw_sip) To: sip:0031*...@sip.xs4all.nl;tag=00-08168-044b6f36-245cd72c7 Via: SIP/2.0/UDP ***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92 Content-Length: 182 v=0 o=cp10 125193221174 125193221174 IN IP4 82.101.62.66 s=SIP Call c=IN IP4 194.109.8.2 t=0 0 m=audio 36984 RTP/AVP 8 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=ptime:20 a=sendrecv - --- (12 headers 10 lines) --- Found RTP audio format 8 Peer audio RTP is at port 194.109.8.2:36984 Found audio description format PCMA for ID 8 Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.109.8.2:36984 -- SIP/*-089ca9b8 is ringing -- SIP/*-089ca9b8 is making progress passing it to IAX2/2104-2287 Scheduling destruction of SIP dialog '740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 82.101.62.99:5060: CANCEL sip:0031**...@sip.xs4all.nl SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport From: ** sip:**...@sip.xs4all.nl;tag=as70e84199 To: sip:0031**...@sip.xs4all.nl Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- However when I dial exactly the same from VoipBuster, I see this instead: -- --- SIP read from 77.72.169.129:5060 --- SIP/2.0 183 Session progress Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport From: * sip:**...@sip.voipbuster.com;tag=as1374705a To: sip:0031**...@sip.voipbuster.com;tag=120113ac4a54a269af9e2c Contact: sip:0031**...@77.72.169.129:5060 Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 162 v=0 o=* 1251932194 1251932194 IN IP4 194.221.62.33 s=SIP Call c=IN IP4 194.221.62.33 t=0 0 m=audio 8958 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 - --- (11 headers 8 lines) --- Found RTP audio format 0 Peer audio RTP is at port 194.221.62.33:8958 Found audio description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.221.62.33:8958 -- SIP/-089dc538 is making progress passing it to IAX2/2104-8077 == Connect attempt from '127.0.0.1' unable to authenticate Scheduling destruction of SIP dialog '1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 77.72.169.129:5060: CANCEL sip:0031**...@sip.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport From: ** sip:***...@sip.voipbuster.com;tag=as1374705a To: sip:0031**...@sip.voipbuster.com Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- As you can see, there are different packets being sent, and in the 2nd case, there is no is ringing message, which is rather irritating... Any suggestions would be appreciated... TIA -- FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voipbuster not ringing, other SIP peers are ringing...
Francesco Peeters wrote: Does anybody else see the same behavior for VoipBuster connections? When I trace one of the other SIP peers, I see it sends this message: -- --- SIP read from 82.101.62.99:5060 --- SIP/2.0 180 Ringing Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl Contact: sip:82.101.62.99:5060 Content-Type: application/sdp CSeq: 103 INVITE From: ** sip:***...@sip.xs4all.nl;tag=as70e84199 Record-Route: sip:82.101.62.115;lr;r2=on;ftag=as70e84199,sip:82.101.63.5;lr;r2=on;ftag=as70e84199 Server: Cirpack/v4.41b (gw_sip) To: sip:0031*...@sip.xs4all.nl;tag=00-08168-044b6f36-245cd72c7 Via: SIP/2.0/UDP ***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92 Content-Length: 182 v=0 o=cp10 125193221174 125193221174 IN IP4 82.101.62.66 s=SIP Call c=IN IP4 194.109.8.2 t=0 0 m=audio 36984 RTP/AVP 8 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=ptime:20 a=sendrecv - --- (12 headers 10 lines) --- Found RTP audio format 8 Peer audio RTP is at port 194.109.8.2:36984 Found audio description format PCMA for ID 8 Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.109.8.2:36984 -- SIP/*-089ca9b8 is ringing -- SIP/*-089ca9b8 is making progress passing it to IAX2/2104-2287 Scheduling destruction of SIP dialog '740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 82.101.62.99:5060: CANCEL sip:0031**...@sip.xs4all.nl SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport From: ** sip:**...@sip.xs4all.nl;tag=as70e84199 To: sip:0031**...@sip.xs4all.nl Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- However when I dial exactly the same from VoipBuster, I see this instead: -- --- SIP read from 77.72.169.129:5060 --- SIP/2.0 183 Session progress Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport From: * sip:**...@sip.voipbuster.com;tag=as1374705a To: sip:0031**...@sip.voipbuster.com;tag=120113ac4a54a269af9e2c Contact: sip:0031**...@77.72.169.129:5060 Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 162 v=0 o=* 1251932194 1251932194 IN IP4 194.221.62.33 s=SIP Call c=IN IP4 194.221.62.33 t=0 0 m=audio 8958 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 - --- (11 headers 8 lines) --- Found RTP audio format 0 Peer audio RTP is at port 194.221.62.33:8958 Found audio description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.221.62.33:8958 -- SIP/-089dc538 is making progress passing it to IAX2/2104-8077 == Connect attempt from '127.0.0.1' unable to authenticate Scheduling destruction of SIP dialog '1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 77.72.169.129:5060: CANCEL sip:0031**...@sip.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport From: ** sip:***...@sip.voipbuster.com;tag=as1374705a To: sip:0031**...@sip.voipbuster.com Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- As you can see, there are different packets being sent, and in the 2nd case, there is no is ringing message, which is rather irritating... Any suggestions would be appreciated... TIA BTW: I am talking about the ringtone the caller should hear... The other side is ringing, and calls are established just fine, but it is very irritating to hear nothing until the call either fails or gets picked up... -- FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk
Re: [asterisk-users] Echo and static on PRI with errors
John F. Ervin wrote: What do you do if you find things sharing interrupts (IRQ 11) in my case with my X100P card. I believe there is some sort of internal audio card in my cheap slow PC. Check the BIOS whether you can: Change the IRQ assignments Disable the extra hardware using the same IRQ Or otherwise try changing the slot it is in... I had very good results in the past swapping card around Good luck! --FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need to find small footprint asterisk platform
Tzafrir Cohen wrote: On Mon, Mar 30, 2009 at 06:20:20PM +0100, Chris Bagnall wrote: One of the more common embedded platforms for Asterisk is the Soekris net5501 (or 4501 if you don't need as much processing power) Agreed. Though, given the Asus eeeBox (1.6Ghz Atom) can be had for almost the same money (Soekris stuff isn't cheap in the UK) and is about the same footprint, it might be worth considering that instead if you don't need ISDN or POTS connectivity. I've done a few Asterisk-based eeeBoxes over the last few weeks and been very impressed with them. In fact, with a netbook I suspect you'd be paying quite a sum for the display. Both in the price and in the heat consumption. Who's talking about netbooks? :-o What screen? --FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN BRI Asterisk 1.4
Lee Wilson wrote: --- On Wed, 14/1/09, Ex Vito ex.vitor...@gmail.com wrote: While I don't know the OpenVOX B200P specifics, some interface cards need you to change physical jumpers in order to acheive NT vs TE, mode. Could that be the case ? -- exvito I've just checked the card and you were right the jumpers had not been changed to NT. I've done this now and also enabled power on one of the ports as this is also mentioned in the manual. However, still neither port comes up on L1 when I connect the router. Once I've changed this jumper setting do I still need to manually change the mISDN.conf file to use NT? When I do mISDN scan/config it is still setting the ports to TE which I then manually edit back to NT. Also, I guess at this point it doesn't matter for L1, but should I be using Point-To-Point or Point-To-Multipoint? Thanks Yes, you would still need to configure mISDN correctly as well! And AFAIK you will need to use PTMP, as that is what the router would expect... -- Francesco Peeters Ubuntu all the way! 1 laptop, 1 server, 1 desktop at home and several servers in different locations ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN BRI Asterisk 1.4
Lee Wilson wrote: Also, I guess at this point it doesn't matter for L1, but should I be using Point-To-Point or Point-To-Multipoint? Thanks Yes, you would still need to configure mISDN correctly as well! And AFAIK you will need to use PTMP, as that is what the router would expect... -- Francesco Peeters Thanks for clarifying I've double-checked that it is running ptmp but still no link lights. Anyone got other suggestions? Regards Lee Are you using an ISDN cross cable? I don't know these cards, but most cards are wired as a DTE type device (TE port like a router or phone) and not a DCE type device (NT box). So you might have Tx-Tx and Rx-Tx instead of Rx-Tx and Tx-Rx... ;-) (Note that ISDN cross cables are definately NOT the same as a CAT5E cross cable!) -- Francesco Peeters Ubuntu all the way! 1 laptop, 1 server, 1 desktop at home and several servers in different locations ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dutch Asterisk mailing list?
Erik de Wild: Tripple-o wrote: What is the most reliable method for Asterisk to detect the Called ID for incoming calls on an analog line in the Netherlands? In Holland you have to pay to receive cid info on the incoming line. If you don't pay at the moment you can start with that. There are 2 ways for a provider to deliver the cid,ETSI en FSK. In Holland (with a couple of other countries) ETSI is used so if you have a phone that only supports FSK the CID will never work. I still have a couple of ETSI - FSK converters catching dust. So if you pay for CID but your phone doesn't support and you have a pot line connected to your Asterisk server I can provide you with a solution for a couple of EUR. If you use the proper card maybe you can adjust the settings so it supports ETSI instead of FSK. I used X100P cards and needed the convertor to get proper CID If the Dutch mailing list starts I will join ;-) Erik de Wild Tripple-o Me three! ;-) -- Francesco Peeters Ubuntu all the way! 1 laptop, 1 server, 1 desktop at home and several servers in different locations ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: [newtech-1] Skype 24 Hour Rolling Analytics
Drew Gibson wrote: SNIP I suspect that this is due to the call billing structure in Europe. They make the North American telcos look positively philanthropic. Yes indeed! Flat rate calling plans? What are those? Flat rate Mobile Internet? non-existant! We pay per minute/SMS/MB on every plan, and the only thing you achieve on a more expensive plan is to pay less per unit, but flat-rate is NON-EXISTANT... It is one of the few things I actually envy my US colleagues for! (Of course, we do have more PTO! G) -- Francesco Peeters Laptop: IBM T43 with Ubuntu Gutsy Gibbon, Workstation Server: P4i65G, 2.4GHz with Ubuntu Gutsy Gibbon, Server Edition Postfix, Dovecot, Mailman, Apache2, Squirrelmail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best alternative for getting prompts recorded.
Steve Edwards wrote: On Wed, 12 Mar 2008, [EMAIL PROTECTED] wrote: Thanks everyone for the reply. Till now we had simple IVR so we recorded it ourself. Now I have a requirement where customer needs a customized message to be played to customer. I am basically looking for some Text to Speech software that would be cost effective (most probably a open source) and would convert Text to Speech. I tried Fetival, but the quality of the sound is not good. Can we improve the sound quality of Festival somehow. Cepstral with Allison is only $30. I did a demo IVR for a potential client and it was hard to tell the TTS bits from the human bits. If I took the time to learn Cepstral's markup language I probably could have fooled myself :) Thanks in advance, Are there any tools like these for Dutch language Asterisk installs?... -- Francesco Peeters no sigs on this machine! :-o ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Please unsubscribe or moderate [EMAIL PROTECTED]
All these repeated list replies with Autoreply: Autoreply: Autoreply: Autoreply:... subjects are irritating at best and debilitating at worst! This makes the list waste bandwidth and my inbox (and the archives too) unreadable! Thx! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN^H^H^H^H^HmISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The downside of Asterisk and least cost routing...
On Fri, May 11, 2007 08:21, Gordon Henderson wrote: On Thu, 10 May 2007, Francesco Peeters (Asterisk) wrote: If you think your ISP is reliable enough then go for it! I've had less ADSL issues last year than ISDN issues! ;-) (And that while ADSL is running over that very ISDN line!) There is a small (and growing!) number of small businesses (and not so small ones either!) who are moving towards using their broadband (typically ADSL in the UK) connection for Telephony - and even installing a 2nd ADSL line just for VoIP. It can work out a lot cheaper than going down the traditional ISDN2/ISDN30 route for a lot of people as a small business expands. I can see that would work out that way, yes! Undfortunately I'll have to pay reconnection fee before I can cancel! :-o I guess that's a country thing - good luck :) I found out that I can even transfer my current main number to my ISP's SIP service for EUR 5 a month... Aside from that they can give me 2 free incoming numbers in the 087 range, and I already have an incoming VoipBuster number in my own areacode... That would give me 4 incoming numbers... The only thing I'd probably lose is the ability to do faxes! So I am going to investigate that further first! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] The downside of Asterisk and least cost routing...
On Fri, May 11, 2007 10:31, Chris Bagnall wrote: There is a small (and growing!) number of small businesses (and not so small ones either!) who are moving towards using their broadband (typically ADSL in the UK) connection for Telephony - and even installing a 2nd ADSL line just for VoIP. Indeed, many of our clients are doing just that. I would, however, strongly recommend against ditching PSTN entirely (in the UK, it's virtually impossible anyway since ADSL requires a PSTN line over which to run) - those PSTN lines are still useful for things like emergency service calls, directory enquiries, etc. etc. In NL you actually can ditch the telephony and keep the ADSL... My ISP even gives emergency access if you transfer your main number to their SIP service. And there still is my cell-phone too! ;-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The downside of Asterisk and least cost routing...
I forgot to pay this month's phone bill, and never noticed until family (the in-laws, who are too cheap to try the cell phone if landline fails, because it is 'more expensive') told me they were unable to reach us... As it turns out, the phone company disconnected us, but because Asterisk routes all outgoing calls in the Netherlands over VoipBuster, I never noticed anything! ;-) If I'd given out my VoipBuster DID, I'd probably still wouldn't know! *ROFLOL* It gives me pause though... Maybe it's time to get rid of my fixed line... ;-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The downside of Asterisk and least cost routing...
On Thu, May 10, 2007 23:44, Gordon Henderson wrote: On Thu, 10 May 2007, Francesco Peeters (Asterisk) wrote: It gives me pause though... Maybe it's time to get rid of my fixed line... ;-) No ;-) needed - I have friends on cable internet with no separate copper phone line now. I'd consider it myself if I weren't tied to having ADSL over my phone line, and as yet there isn't a way to separate them (in the UK) In NL there is... ;-) Especially interesting as I have ISDN, which is almost twice as expensive... So I am really going to look in to it... I'd save about EUR 20,00 per month that way! Undfortunately I'll have to pay reconnection fee before I can cancel! :-o -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any other softPBX like Asterisk?
On Fri, May 11, 2007 07:34, Armin Schindler wrote: On Thu, 10 May 2007, Crazy Boy wrote: Hi Friends, Can anybody tell me other softPBX softwares like Asterisk? - OpenPBX - Freeswitch Or try Googling for something like 'open source pbx'... Sheesh! :-o -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] freepbx - DB Error messages...
On Sat, March 24, 2007 19:10, Bruce Reeves wrote: You might get a faster response on freepbx/amp mailing list. On 3/24/07, Francesco Peeters (Asterisk) [EMAIL PROTECTED] wrote: SNIP Just an update: Still have NOT been approved for either the mailing list *or* the forum! I am pretty disappointed in the moderators! If you take up the responsibility to moderate a list or forum you have to make sure you respond promptly, especially if the list or forum (or both) require moderator approval before a user-account is activated! (And no, my original answer has not been answered yet either!) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 VIA EPIA V8000 - 256 MB - * 1.2.4 - mISDN, but still no freePBX 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error in FreePBX
On Thu, March 29, 2007 19:36, Carlos Jerónimo wrote: Hi Steve, your sugestion is correct, but i registed 2 times in FreePbx foruns this week, and my login is inactive yet. In the mail i receive this msg: Welcome to FreePBX Forums Forums Please keep this email for your records. Your account information is as follows: Your account is currently inactive, the administrator of the board will need to activate it before you can log in. You will receive another email when this has occured. Same here... Been waiting a week since my last attempt, but still nothing!... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] freepbx - DB Error messages...
Hi all, I am probably missing something ultimately obvious, but I have a problem configuring freepbx... Using Edgy Eft with the cvs freePBX 2.2.1 and followed the Ubuntu installation guide on freepbx.org. System pxe-boots from a server with NFS root on same Using * 1.2 current (from source, not .deb's) Using mISDN-streams (from source, not .deb's) Using freePBX-2.2.1 (from source, not .deb's) Installed everything, and mISDN and * load just fine amportal start works fine as well However I keep getting DB Error's in the GUI... The syslog gives two separate errors: 1) Error 127 when reading table ./asterisk/whatever 2) Table is crashed and needs to be repaired I created a special mysql user for * and did an PERMIT ALL PRIVILEGES on the mysql databases When I log in to mysql as root and do 'SELECT username FROM ampusers ORDER BY username' I get the record list. When I do the same as the * user, I get the 'Table is crashed, blablabla' line. I tried changing the login user for freepbx (ampdbuser) to root, but that doesn't help either, as I keep getting the 127 error... Googling wasn't very helpful, and the freepbx forum admins still haven't approved my account, so I thought I'd try here... Any help appreciated! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with Hamlet ISDN PCI card(Cologne Chipset)
On Sat, March 24, 2007 11:54, Mauro Zanin wrote: Hi everybody I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded software. I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN CARD in a normal Italian EUROISDN installation. The * works fine except for the ISDN CARD. It is always Channel D down, but if a Call comes in, it works perfectly for some time, both inbound and outbound. It prompts Channel D UP! If I disconnect the NT+ termination the Channel D goes down at once. Did I make something wrong? Not really... It's a bristuff quirk... It doesn't gracefully handle the forced D-channel down that most European ISDN operators implement. That is why I switched to testing vISDN, but that has been stagnant for over half a year without any fixes for a few very annoying bugs, because the programmer dedicated all his time to rewriting the vGSM part... I am now testing mISDN as someone on the vISDN list mentioned that it's chan_misdn voice support had greatly improved... The only way I can *somewhat* keep bristuff working without contacting the ISDN carrier to turn on the D channel permanently is by initiation a 100ms outbound call every minute using the manager interface... (Yes, a very ugly kludge indeed, but I do not want permanent channel up, as I want to be able to test everything in a normal environment, as I am planning to install this in other location too once I have a stable, reliable environment) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX softphones
On Wed, October 18, 2006 19:03, Paul Gaffney wrote: Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm looking for a NAT-friendly solution and my SIP phones are good but not dependable. Neil Neil, www.asteriskguru.com http://www.asteriskguru.com/ lists a few of them. Try IDEFISK. Paul Gaffney LANStatus,LLC I personally like DIAX on for Windows users. Haven't yet found an IAX phone I like on Linux... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX softphones
On Wed, October 18, 2006 21:07, Guillermo Salas M. wrote: On Wed, 2006-10-18 at 20:08 +0200, Francesco Peeters (Asterisk) wrote: On Wed, October 18, 2006 19:03, Paul Gaffney wrote: Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm looking for a NAT-friendly solution and my SIP phones are good but not dependable. Neil Neil, www.asteriskguru.com http://www.asteriskguru.com/ lists a few of them. Try IDEFISK. Paul Gaffney LANStatus,LLC I personally like DIAX on for Windows users. Haven't yet found an IAX phone I like on Linux... Kiax works great with Gnome, KDE or Xfce. -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 I'll try that later, thanks! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
On Tue, September 26, 2006 22:21, Barry Fawthrop wrote: Hi all I didn't change anything that's my point It has be running and working just fine then at 4:32 pm yesterday I could not make or recieve VoIP calls via our VoIP Provider They say the Invite packet was being rejected and thus there was no real connection even though SIP SHOW PEERS has us registered They also say it's due to the Sonicwall which has changed port assignments and thus blocking ports. I see in the Sonicwall log UDP Packet Dropped with the Providers IP Address but it talks about port 36612 which is not SIP They say along with the log that SIP is using 36612 why when all the VoIP SIP setting are enabled/configured and SIP is packet forwarded to the Asterisk Box due to Sonicwall NAT Now I'm trying to find out why and how to correct this. Thanks all Barry SonicWALL Enhanced has an option called 'Persistent NAT'... Is it turned on? -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any Hardphone with VPNClient embedded?
On Mon, September 4, 2006 16:55, Cory Andrews said: Please be aware that from a future support standpoint, you may be a bit limited with Zultys. Their future seems very uncertain they have recently just about ceased operations and let the majority of their employees go. Cory J Andrews voice - 800.398.VoIP X3402 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Leo Ann Boon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 04, 2006 10:35 AM Subject: Re: [asterisk-users] Any Hardphone with VPNClient embedded? Marco Mouta wrote: Hi all, Does any of you knows an Hardphone with VPN client embedded? Take a look at Zultys SIP phones. VPN enabled. www.zultys.com As I too am interested in IPsec capable hardphones (or ATA's), do you have a suggestion what to look at instead? I mean: It's nice to say the company may not be around for long, but if there's no alternative, what choice does one have? TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoipNow 1.2.0 Beta
On Mon, July 31, 2006 21:44, Tom said: At 02:21 PM 7/31/2006, you wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Senad Jordanovic wrote: [EMAIL PROTECTED] wrote: Tom Vile wrote: Did you look on the site? http://www.4psa.com/products/voipnow/demo.php Does above means that the license for voipnow need to be paid to packet 8 as well? http://biz.yahoo.com/prnews/060613/sftu062.html Senad Hate replying on my post but what a heck!!! My understanding is that ANY hosted IP PBX coded in any object oriented programming language is falling under the above mentioned patent. Anyone has any thoughts on this? Another reason not to do business in the USA! Any good suggestions on where to buy rack space in a country that is not honoring stupid US patent law and has great and secure Internet connections? Tom Ehrm... Russia, China... You could also try several European countries, such as the Netherlands, Luxembourg, Switzerland... I just have mine at home... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 or more ISDN cards: which comes first ??
On Thu, June 29, 2006 8:40, Stefan-Michael. Guenther (in-put GbR) said: Hello, I have setup an asterisk server with two EICON cards, a 4BRI and 2FX. How do I know, which card is the first, so that I can setup capi.conf with the right entries? Thanks for your help, lspci should tell you... -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Wed, June 28, 2006 10:14, [EMAIL PROTECTED] said: Well, look at it this way: if you get the working, you can buy one of those tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard and a ethernet port. Run Linux off a CF card and have it setup to *only* interface with Skype and Asterisk. Basically, make a Skype ATA, but it would convert Skype to SIP. I think that could still be considered an ATA, right? Or a gateway at least. Since you can make a Skype account for free and can (for right now) make US and Canada LD calls for free, I think the cost and time to make them would be worth it. :) And if you figure out a good price for them, people might even buy them from you Undrhil Another advantage is that you can reach all those people who have Skype and are not willing to try Voipbuster or similar SIP based providers, and tell them that SIP/IAX/Asterisk *is* the better solution, because they cannot do the same with Skype the other way round! ;-p -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP
On Tue, June 27, 2006 0:26, shadowym said: They have been talking about this for awhile. If you look at the real time and embedded operating system world they have not really done so well over the many years they have been trying. Just throwing money at the problem has never worked for them in the past either. Perhaps because people expect devices like that to Just Work(tm), something Embedded Linux is better known for than Embedded Windows is?... The Asterisk community has nothing to worry about in the near term if ever IMHO. Unless they buy Digium... That'd give them a serious amount of code to obfuscate and hide in closed source products! ;-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Mon, June 26, 2006 20:06, Brian Capouch said: Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. Let them talk. What's it hurt the rest of us? It is more a question of netiquette... If you're on an English mailinglist, you should speak English (Not attacking Josué and Marco, just answering the question here). It is not only more productive (If you keep to English, more people understand and can contribute to *and* profit from the discussion), but speaking a different language not spoken by the majority on list is generally considered akin whispering in company: not quite rude, but also not-done... We have seen the wages of tortured English sometimes unleashed on the list. If they're getting the job done, I say hit the Delete button and get on with your life. You can hit the delete button for bad English too, you know! ;-) If 80% of the list traffic were in foreign languages, then I would say we would have an issue. Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Mon, June 26, 2006 21:39, Brian Capouch said: Francesco Peeters (Asterisk) wrote: Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? B. Ningunas quejas aquí... Apenas una explicación en el 'netiquette' --FP ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] David Choo/eServices/eSpore is overseas
On Mon, June 12, 2006 4:37, David Choo said: I will be out of the office starting 12/06/2006 and will not return until 17/06/2006. Dear Sir / Mdm, I'm currently travelling. During this period of time, I have minimal access to internet and email. As such, please be aware that I might not be able to reply to your queries promptly. I apologise for the inconvenience caused. SNIP Tongue mode='in cheek' That is good to know! We will start monitoring your residence until we find an opportune moment to enter. We will then lend a hand in (re)moving the most precious of your things to a new address... /Tongue (Sorry, couldn't help myself!) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: GXP-2000 (steer clear)
On Wed, June 7, 2006 14:09, Louis-David Mitterrand said: On Tue, Jun 06, 2006 at 11:26:20PM -0400, Daniel Salama wrote: Well, these are encouraging words :) You're basically telling me that I should tell my client to buy other phones. I agree that you cannot compare these phones with Cisco or Polycom. After all, like you said, what do you expect for under $90. However, the fact is that my client just recently invested in these and it will be hard, if not impossible, for me to tell my client to swap them for Polycoms or something else at a much higher cost. I have heard complaints from my client about the speakerphone and they are now, I guess, getting used to picking up the handset :). I have heard any echo problems so far. What bothers me the most is that the phone stops working often (multiple times per day). By this I mean that my client won't be able to dial anything successfully. As soon as 3 or 4 digits are entered, they get a fast busy. To solve it, they need to reboot it. It sounds as if these phones were running Windows instead of Linux :) Anyway, what firmware did you use that solved so many of your problems? I've only had bad experiences with these phones and steer clear of them. In the same price range you can now get the Thomson ST-2030 or Polycom 430 for a much, much better user experience. Where do you purchase the Thomson or Polycoms for a comparable price as the GXP2000? I'd like to buy an ST2030 or 430 for under EUR 90 myself too! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] registration at Voipbuster times out
On Mon, May 29, 2006 16:20, Remko Muis said:
Hi Steve Attilla,
Thanks for the quick replies!!
Attilla: your suggestion sounds promising, since I know my system clock is
not too accurate. But that is the reason I use the network time protocol
daemon. Time and date settings are now correct.
Steve: your question about pinging the sip-proxy servers hits the nail on
its head: I can't, even though the names resolve to ip-addresses, and I
can
ping lots of other machines in the outside world. But why?
I tried your second suggestion, but to no avail. My dial statements were:
exten = _0[12345789],1,Dial,SIP/voipbuster-out/0031${EXTEN:1}
exten = _0[12345789],2,Congestion
exten = _XXX,1,Dial,SIP/voipbuster-out/0031[b]10[/b]${EXTEN}
exten = _XXX,2,Congestion
Replacing voipbuster-out with username:[EMAIL PROTECTED] does
not
help.
However, I did not really expect so, since the registration timeout errors
occur while Asterisk executes chan_sip.c. I would think that registration
fails independently of any wrong settings in extensions.conf.
Anyway, the s in the Contact-line does look suspect to me, since I have a
voip-in number for Voipbuster, and I read on the voip-info pages that the
s
extension is is used when there is no known called number in the context
used.
Being an Asterisk-newbie, I appreciate your replies, but further
suggestions
even more ...
Remko
Remko,
What IP's do you get returned for sip.voipbuster.com?
Do you use UU-net's DNS servers? If so, you might try using different
servers, as I have had some weird experiences with their DNS servers in
the past.
Have you tried trace-routing to the server to see where it breaks?
I am using voipbuster as well, and am usually able to connect just fine...
Good luck!
--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0
AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN
2 Sweex HFC-PCI cards
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Re: [Asterisk-Users] USB headsets?
On Wed, May 24, 2006 10:16, El Flynn said: [EMAIL PROTECTED] wrote: We have some laptop soundcards that are really bad and I would be glad if you could share your experiences when changing to a USB headset instead of using the built in soundcard in your computer. Well, IMO if the soundcards are already crap to start out with, there's no way a fancy-schmancy USB headset -- or any other headset, for that matter -- will sound good when plugged in to the laptop. Because, remember, it's the soundcard that generates the audio and sends it out the heaphone port. Flynn Not true... The USB driver generates a stream which is sent to the USB headset, which then makes it in to sound. The soundcard has nothing to do with that. (Otherwise, how can they work on soundcard-less machines, like my old DELL?) If the source is bad (for instance the crappy soundcard) then the USB headset won't make a difference. I have a logitech USB headset and a labtec USB headset, and love both. The Logitech has better audio though, so when using it to listen to music, etc., you'd better be looking at something similar. Or get a USB audio-device with input/output jacks, so you can plug in whatever you want... -- Francesco ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] USB headsets?
On Wed, May 24, 2006 10:40, [EMAIL PROTECTED] said: I have a logitech USB headset and a labtec USB headset, and love both. The Logitech has better audio though, so when using it to listen to music, etc., you'd better be looking at something similar. Or get a USB audio-device with input/output jacks, so you can plug in whatever you want... What model headsets (name/number) do you have? Any recomendations on the USB audio-devices? Thanks! Regards, Jan I have the direct predecessor of the current Logitech 350. I never tried it on Linux, but it works great on OS/X and Win** AFAIK Labtec no longer sells USB headsets... Good luck! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delay when ringing internal extensions on incoming zap call
Derek Lee-Wo schreef: do... In any case, removing the fax detect seems like it should help. I commented out the line and it works great. Two concerns though: - I edited the extensions_additional.conf file, but my fear s that it will get overwritten if I make further changes via AMP or upgrade. Is there a way to remove the NVFaxDetect() by only editing *_custom.conf files? - According to the documentation I was able to find, the 0 in NVFaxDetect(0) means to not wait, but obviously it still waits at least one ring. Going to AMP, Setup - General - Extension of fax machine for receiving faxes = disabled *should* disable fax detection by causing it to use a different branch of the AMP macro's... HTH! -- Francesco Peeters ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoipBuster issues?
Hi All, Any VoipBuster SIP users on this list that'd be willing to test VoipBuster outbound VoIP to PSTN? All numbers I tried from my (*) server are supposedly being connected, but no phone rings! Also their new WebStart function doesn't cause my phone to ring either... TIA! -- Francesco Peeters ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need a Service that allows me to call Toll Free Outbound numbers
Bob's Leaky News Service schreef: Simple as that please email me direct. [EMAIL PROTECTED] Also looking for a U.S. DID provider as well as orig provider. FWD (FreeWorld Dialup) will allow Toll Free outbound... So will some of the Finarea services (check the wiki for their various services. According to http://www.voip-info.org/wiki/view/Finarea+SA you get free outbound USA calls (not quite free, as you have to pay a small amount to add call credits to your account which have to be replenished every 120 days) from: internetcalls; sipdiscount; voipcheap; voipdiscount and voipstunt) HTH! -- Francesco Peeters PIII-450 - 512 MB RAM - 2x HFC-PCI - BRIstuff Florz patch AMD Duron 1GHz - 512 MB RAM, 2x HFC-PCI - vISDN ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SSH from System() ?...
Hi, I would like to execute a command on a different system using ssh. When I execute the command from the CLI on the asterisk machine, it works fine (I set up RSA keys on both sides) When I execute the same command from System() inside the dialplan, the log shows it is being executed, and a session is established, but the remote host never receives the command. I *think* it has to do with the command shell environment in which the system command is opened... Any suggestions on how to set this up would be appreciated... -- Francesco Peeters PIII-450, 512 MB, 2x HFC-PCI, BRIstuff Florz patch AMD Duron 1GHz, 512 MB, 2x HFC-PCI, vISDN ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SSH from System() ?...
Francesco Peeters schreef: Hi, I would like to execute a command on a different system using ssh. When I execute the command from the CLI on the asterisk machine, it works fine (I set up RSA keys on both sides) When I execute the same command from System() inside the dialplan, the log shows it is being executed, and a session is established, but the remote host never receives the command. I *think* it has to do with the command shell environment in which the system command is opened... Any suggestions on how to set this up would be appreciated... Never mind, I had wrong permissions on the authorized_keys file on the target machine! -- Francesco Peeters PIII-450, 512 MB, 2x HFC-PCI, BRIstuff Florz patch AMD Duron 1GHz, 512 MB, 2x HFC-PCI, vISDN ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [EMAIL PROTECTED] and channel announcement
I have been pondering the following... Voipbuster used to announce the cost of the call, but the new SIP servers do NOT. Because there's the choice between free (VoipBuster) and non-free (ADSL), I'd like to let the user know which one is actually being used by announcing it before the actual call gets connected, ie immediately after the channel proceeds from setup to actual ringing... Is there any way of making this happen, and preferrably with as little change to the [EMAIL PROTECTED] macro's as possible? Any ideas would be appreciated... -- Francesco Peeters No sigs on this machine yet ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call terminated after 60 seconds
On Fri, March 24, 2006 12:01, Asterisk said:
Hello,
I switched from my PSTN provider to a voip provider. (Voicedata in
the Netherlands)
From the moment i switched all inbound calls are terminated after
aproximatly 1 minute.
The provider tells me it's not their issue since I have no other
configuration than all their other users.
What can I do.
I removed all asterisk functionality by forwarding the inboud call
directly to a local phone
; Inbound voicedata context
;
[from-voicedata]
exten = ${VOICEDATACIDNUM},1,NoOp(From Voicedata)
exten = ${VOICEDATACIDNUM},n,Dial(SIP/2200,45,tr)
; end of context
Regards,
Andre Vink
Check whether your firewall has a fixed UDP timeout set at 60 seconds...
That solved my problem... ;-)
(Together with activating SIP/VoIP support)
--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0
AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN
2 Sweex HFC-PCI cards
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Re: [Asterisk-Users] Zap--IAX codec?
On Tue, March 21, 2006 16:51, Mimmus said: Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial(Zap/2-1, IAX2/215|20|TtwW) in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) -- Format for call is ulaw -- IAX2/215-33 is ringing -- IAX2/215-33 answered Zap/2-1 Why I have 'Format for call is ulaw'? I'd like to have alaw but keep ulaw to accomodate errors in various configurations (if any, not here!). EuroISDN uses uLaw, so Asterisk does as well, because it doesn't need to do transcoding then... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap--IAX codec?
On Wed, March 22, 2006 0:06, Steve Kennedy said: On Tue, Mar 21, 2006 at 10:57:06PM +0100, Francesco Peeters (Asterisk) wrote: Why I have 'Format for call is ulaw'? I'd like to have alaw but keep ulaw to accomodate errors in various configurations (if any, not here!). EuroISDN uses uLaw, so Asterisk does as well, because it doesn't need to do transcoding then... Err,, uLaw is used by North America (as in U(s)Law ;) aLaw is used in Europe and other sensible areas. Steve Oops, you're right... my Bad! Sorry! (It's been a very long and tiresome day yesterday... I should have just kept my mouth shut!) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX choppy sound
On Thu, March 16, 2006 12:08, Stojan Sljivic - GDS said: Hi, Does anyone know what would be acceptable RTT. Is 200ms OK? Regards, Stojan Sljivic When any of my VPN tunnels get over 100ms I start to get worried! Avg speeds on the tunnels are below 45 ms... I guess it depends on the level of quality you're used to tho! (As well a how far aprt the networks are... Mine are all in the same country...) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 codec licencing
On Thu, March 16, 2006 22:38, rnacharya said: Hi.., we have two asterisk server interconnected to each other through IAX2 trunk in two separate office. with this bellow configuration do we need to have Licensing for using G729 codec Office A T1 - Astrisk TE05PIAX2Astrisk Box -2 | | | | | | EPBX-1 EPBX-2 | | | | Telephone Telephone Thanks. Rudra. Your information is too summary to be able to tell... If EPBX-1 and -2 do G729, and the (*) servers only pass it, then you won't need additional licenses. Unless the (*) servers need to handle voicemail from either side. If the (*) servers have to transcode from any other codec *OR* from analog or ISDN (uLaw/aLaw) then you'll need licenses allright... Your best bet may be to contact Digium and give them all the details they need to determine the correct # of licenses... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 + Sonicwall
On Fri, March 10, 2006 14:49, Dr. Michael J. Chudobiak said: I've found that inbound IAX2 calls don't work reliably (i.e., I get a busy tone) unless I enable Use Consistent NAT in the Sonicwall. This feature is poorly documented by Sonicwall, so I thought I'd pass it along. I've used the iaxcomm softphone and a snom 200 behind serveral different sonicwalls over the past year or so without any problem. The sonicwall should not be a problem for iax calls at all. I think the problem occurs when an Asterisk server inside the firewall tries to register multiple DIDs with one IAX2 provider outside the firewall. The Asterisk server worked fine when it was connected outside the firewall. The Sonicwall TZ170s do handle SIP transformations very nicely, though, if your Asterisk server is outside the firewall. If the persistent NAT is not enabled, the SonicWALL is allowed to change the NATted (source) portaddress. I can imagine that changing the port on an IAX2 connection can cause problems on inbound sessions. When Persistent NAT is on, the SonicWALL is told to use the same portnumber as the original request from the LAN based machine. This can cause problems if you have multiple machines connecting to the same remote resource as there is a 1 in (approc) 64k chance per connected machine that it uses the same port number as another machine that already has a session up. The chance of it causing a screw up are small enough to be able to have it turned on, as I have. I am not sure what the default is for new machines, but I know older machines that were upgraded to newer firmware will be off by default... HTH! (PS: It's not the only thing poorly documented by SNWL... They unfortunately have a history of poor documentation! It *does* keep their support agents working though, so I guess that's something! G) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 + Sonicwall
On Fri, March 10, 2006 18:56, Rich Adamson said: Hi all, I currently have an Asterisk test server behind a TZ170 Sonicwall firewall / NAT box, with several DIDs. I've found that inbound IAX2 calls don't work reliably (i.e., I get a busy tone) unless I enable Use Consistent NAT in the Sonicwall. This feature is poorly documented by Sonicwall, so I thought I'd pass it along. Has anyone else run into this, or figured out the rationale for it? I've used the iaxcomm softphone and a snom 200 behind serveral different sonicwalls over the past year or so without any problem. The sonicwall should not be a problem for iax calls at all. OK apart of my beleive that sonicwall is a piece of crap (personal), try to do a port forwarding for the IAX port (4569) I don't have a sonicwall here to test with. The ones that I was referring to are production units in Schools and Banks in the Midwest, and I was using iaxcomm (inside) to access asterisk (outside on a registered IP). In general terms, the customers that have them are very satisfied with them, but most don't have to deal with their tech support. The state of South Dakota has standardized on the sonicwall stuff and really have not had any significant issues (they have had some minor ones though). We've also configured some of those boxes to port forward (to the inside) for various functions, but none with udp 4569. Never had any issues other then making sure another rule isn't higher in the chain doesn't block first. The majority of those sonicwall boxes have been the larger units intended for small business use. No experience here with the small entry-level boxes. Our company does not resell any hardware or software, and we have no association with sonicwall whatsoever. SonicWALLs are targeted at SMB and above. They don't really have 'entry-level' boxes. I currently have a TZ170 and SonicPoint combo, and don't have any serious issues with it. There are some minor issues, but there's no product I know of that doesn't have some issues... On the whole I am pretty satisfied with them and would recommend them to others. (BTW: I do not sell SonicWALLs nor do I work for the company) Other people like Zyxel's, which I think are crap... To each their own! ;-) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Fwd: Over 40 destinations for FREE!]
Just in my Inbox: Original Message Subject: Over 40 destinations for FREE! From:[EMAIL PROTECTED] [EMAIL PROTECTED] Date:Thu, March 2, 2006 17:40 To: -- Dear Voip-Fan, From the makers of Voipbuster: http://www.internetcalls.com Over 40 FREE destinations, PLUS free VoipIn number AND Call Forwarding! For more rates, click here: http://www.internetcalls.com/en/rates.html Kindest regards, The VoipBuster Team If you want to be removed from our mailing list click here: http://www.voipbuster.com/en/feedback.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Fwd: Over 40 destinations for FREE!]
On Thu, March 2, 2006 18:26, trixter aka Bret McDanel said: On Thu, 2006-03-02 at 17:51 +0100, Francesco Peeters (Asterisk) wrote: Just in my Inbox: From the makers of Voipbuster: http://www.internetcalls.com Over 40 FREE destinations, PLUS free VoipIn number AND Call Forwarding! Finerea has sipdiscount.com which also is offering the same deal. it appears they have peaked now and are mailing everyone off all their family of sites. I got one a while back for um something other than voipbuster I forget which of the 10 companies they operate (all basically the same deal). sipdiscount still makes you sign up with their stupid windows client but it freely gives you the sip settings so you dont have to guess if its sip.voipstunt.com or connserver.whatever or ... my guess is they are deprecating the other sites soon becuase they seem to really want to push internetcalls.com ... With all the sites integrated in to a single set of servers, and apparently the only difference between all service being the username, I'll stick with VoipBuster as long as I have credit... (My 120 days are passed, but my account and credit are still there... Maybe because I purchased before the expiration bit came in to play. (Might have to do with the many laws in Europe that do not allow for conditions to be changed *after* the purchase has been confirmed)) When the credit is almost gone, I'll check the situation again... ;-) The only thing I really miss is the free US calls... As long as most of Europe is free (esp The Netherlands and - in lesser extend - Belgium (Which was recently added back to VB!)) I am content... BTW: internetcalls.com has (currently) more free destinations than both VB and SD!... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Disconnects - or ARE they?
On Wed, February 15, 2006 22:35, Brent Torrenga said: I have one use on our PBX who has been experiencing seemingly random disconnects. The user is on the same LAN as everyone else, using the same type of phone (79XX loaded with SIP firmware) as everyone else. He had some disconnects a few weeks ago, I suspected the phone, so I swapped his with mine. I have since not had issues with his old phone, however, he has had issues using mine. So, the problem seems to be not with the phone, but with his station. I started thinking maybe the cable is bad. I checked the network stats on his 79XX, and never see any receive errors - perfect network performance. Also, the CLI has no indication of an error whenever a disconnect occurs, it just looks like a normal hangup of the Zap channel (TDM400P). The ONLY difference between this user and everyone else is his extremely loud talking. When I run ztmonitor it is obvious that he simply pegs the meter. Either it reads peaked out or silence, whether he is speaking or being quiet. Is it entirely possible that he is driving the Zap channel so hard that it either hangs up or causes the telco CO to hang up the channel? Is there something else I should look at that might indicate what the problem is? I am kinda pulling my hair out on this one, any help or suggestions would be appreciated. LOL... You could try to explain that he doesn't need to shout to the person on the other side, that the telephone transmits the sound by wire, and not by air, so he doesn't need to shout to be heard on the other side! ;-) But seriously, I am really curious whether there is a connection between voice volume and disconnects... Please do keep us informed... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with ZAPHFC: internal S0 hangs when hanging up
On Tue, February 7, 2006 9:53, Sven Fischer said: Am Dienstag, 7. Februar 2006 09:38 schrieb Sven Fischer: Hello all, if I try to call from one phone on the internal S0 to another on the same S0 using zaphfc, the bus is hung up. The called phone is ringing, but I can't talk from one phone to the other. The error I get is: -- Executing Dial(Zap/2-1, ZAP/1/55|15|tr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1/55 -- Channel 0/1, span 1 got hangup, cause 42 -- Zap/1-1 is circuit-busy -- Hungup 'Zap/1-1' The called phone is still ringing, if I have hung up the calling phone. I have to restart asterisk to get things going again. Calling from SIP to the phones and calling from phones to external ISDN is working fine. Okay, further investigations show that if I connect just one phone to the NTBA, everything seems to work fine. If I plug in the second phone, the communication fails. Each phone works if plugged in on it's own into the NTBA. Termination in the NTBA should be activated, the switches are on. Where should I look for errors? Can it be a termination problem if every phone works on it's own? Sven Is the card set up for multipoint use? (BRI_NET_PTMP) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1 ISDN BRI to IAX2/SIP... (*) best tool or?...
On Tue, February 7, 2006 11:16, Peer Oliver Schmidt said: Francesco Peeters (Asterisk) schrieb: They have several ISDN BRI connections, most of which will be dropped. Only one will be retained, for 2 reasons: 1) It has the ADSL link 2) The number has been the main contact number for over 20 years. In germany you could move that number to a VoIP provider and use it from the main office direct. Then you won't need an asterisk in the remote location. Over here we can as well, but that requires cancelling the line it is on. That would mean we'd also lose the ADSL, and that would mean paying a penalty, paying connect fees all over again and then restart the entire provisioning circus all over again... My question is whether there are any tools better suited for this than an old banger (AMD 800 MHz) PC with a HFC-PCI card and (*) relaying (switch) the incoming calls to the central box. Should be plenty enough. I am running a PII-400 with a AVM C4 connected to two ISDN-ports and have another IAX connection to a customers site. Works fine. I have a PII-450 at home with 2 HFC-PCI cards (1 TE, 1 NT) with a few ISDN-DECT phones and a few IAX phones, which runs great. The only drawback is that starting AGI scripts takes a bit, so in and out bound calls take a bit longer to connect (10-20 seconds...) What I *also* would like to know is whether there's tools that people think would be better suited for this... IMHO a simple (*) box is the cheapest solution available, but I am always interested in novel ideas... ;-) -- F Peeters PII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1 ISDN BRI to IAX2/SIP... (*) best tool or?...
I have a question, I have to provide a solution for an office that will be almost abandoned, and there will be one or sometimes two persons 2 days a week. The main number however should be preserved. They have several ISDN BRI connections, most of which will be dropped. Only one will be retained, for 2 reasons: 1) It has the ADSL link 2) The number has been the main contact number for over 20 years. What we are looking for is to put a single SIP phone in the office, and have it connect back to an (*) server in the central office, where all other servers are located as well. In the remote office a single machine should be placed to terminate the BRI connection and relay it to the (*) server in the central office. That way the old number can be retained and an active phone can pick up the line as necessary. The preferred protocol to use would be IAX2, obviously. My question is whether there are any tools better suited for this than an old banger (AMD 800 MHz) PC with a HFC-PCI card and (*) relaying (switch) the incoming calls to the central box. (No intelligence there, no AGI scripts, just encode and transmit. Also no phones would need to be logged in to that machine, and outbound calling would only take place in very rare cases when the lines *and* VOIP connections at the central site are all congested...) TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Euro-ISDN
On Wed, February 1, 2006 22:12, Armin Schindler said: On Wed, 1 Feb 2006, Aldo Bergamini wrote: [EMAIL PROTECTED] is believed to have said: chan_capi does not set the NT-mode. Your cards driver need to do that. E.g. for Eicon DIVA Server cards, you just set the '-x' option with divactrl or set NT-mode in the config wizard. chan_capi does not (need) to know anything about what protocol the card is doing. CAPI is independent here. Ok. Anyway, if the card is set to NT mode, you should specify ntmode=yes in the capi.conf to tell chan_capi to handle the progress better (get progress tones). Fine! One last related subpoint: while Eicon Diva cards have their own setup application, is there anything standard to control the basic setup of generic HFC-S cards? (something similar to the ztconfig tool for analog cards) Sorry, I cannot answer that one. I don't know enough about these cards and their drivers. With BRIstuff you get to use ztcfg, etc. Cannot say anything about mISDN, CAPI... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
On Fri, February 3, 2006 0:44, Imran Ahmed said: Step 3 The Iax client heve to send some other DTMF to the IVR. How is the IVR still involved if the call has been transferred into a conference room? The IVR records the conversation between the other partecipant to the conference and wait '#' to stop recording and a '1' to save the file. may or may not work, try at your own risk: 1) Use a sip soft phone and set the dtmf mode = inband. 2) In asterisk set the dtmf mode for that soft phone to be rfc2833 or info. (this is done so that asterisk ignores the inband dtmf on the sip channel). 3) Design your dialplan such that asterisk should not depend on dtmf from the sip call. ex: exten xxx, 1, dial(zap/g/client_number) //on answer directed to conference room exten xxx, 2, dial(zap/g/ivr_number) //on answer directed to conference room. exten xxx, 3, meetme(conference room) once the sip call is in the conference then the ivr will detect dtmf from the audio data. Note that before the sip call is in a conference dtmf will not be detectable by the ivr or asterisk, and Ofcourse, this is not tested and only a test can confirm if it works. drawbacks: dtmf will not be available to ivr until your call is in conference. asterisk will never see any dtmf (which should be okay in this specific case). dtmf tones are not squelched so the other user in the conference will hear dtmf tones. Imran What I find strange is that the meetme IVR participant *does* hear DTMF from the ZAP channel, but not from the IAX2 channel... There shouldn't be a per channel difference in how dtmf is handled in meetme, right?... Do you know whether the IAX2 dtmf is intercepted by meetme and handled internally? If so you might be able to workaround by using SendDTMF() in your meetme dialplan... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Leftover sound on isdn modem channel
On Wed, February 1, 2006 7:51, Marnus van Niekerk said: Hi, I have a strange problem on some isdn modem channels. (* 1.0.9 / chan_modem / 2xHFC-S cards). Everything works fine but when the 2nd (and 3rd etc..) call comes in and * answers and there is about a 1/2 second of sound from the previous call (ivr) before the sound from the new call is heard. It just sounds bad and is quite annoying. I am assuming this is sound that is still in a buffer in * or on the modem but can not find any way to get rid of it. I considered muting the channel for 0.75s after answering but could not find a way to do that either. Any suggestions? What ISDN driver set are you using? (Zap/Bristuff/vISDN/mISDN/CAPI?) I see (hear!) the same, but only when using vISDN, bot BRIstuff (haven't tried mSIDN/CAPI) -- Francesco Peeters PalmOS user since Pilot1000 Tungsten|T3 owner, still learning new DateBK5 tricks every day! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
On Wed, February 1, 2006 12:07, Accursio Avona said: Imran Ahmed wrote: Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. I donot know if IaxComm has inband dtmf mode available, if so enable it and see if it works. Someone can suggest me a Iax softphone with inband dtmf mode available ?? Thank's in advance AFAIK there's no DTMF option in IAX2... IAX always sends DTMF inline, eliminating the confusion often found with SIP. http://www.voip-info.org/wiki-IAX -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
On Wed, February 1, 2006 15:04, Accursio Avona said: Francesco Peeters (Asterisk) wrote: SNIP AFAIK there's no DTMF option in IAX2... IAX always sends DTMF inline, eliminating the confusion often found with SIP. http://www.voip-info.org/wiki-IAX If so, wy the IVR does not hear the dtmf sended by the iax client and it hear that one sendee by the zap channel? Could it be a meetme problem? and if so what can i do? Thank yuo very much for any help. Accursio Avona Are you sure it *is* sending DTMF in the first place? (Just trying to find a logical place to start here...) I do not use meetme, but when I use idefisk, my (*) server recognizes the DTMF. Have you tried whether the IAXCOMM DTMF is recognized OUTSIDE meetme? -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipbuster incoming
On Tue, January 31, 2006 14:35, bails said: Hi all, Some friends of mine have an asterisk box which they use for outgoing IAX2 via voipbuster.com. They have been told that they now have an incoming number 0044117*** The thing is I cant seem to get any debug info on the incoming. I have tried both sip and IAX trunks but dont see any incoming info. Anyone have any idea what protocol voipbuster use for incoming calls?? Thanks in advance VB incoming ONLY works with SIP, not IAX2, which will be obsoleted shortly anyway. Incoming context will be the default SIP inbound context Incoming DID will be VB username My (working!) config: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ilbc allow=gsm allow=g726 allow=speex allow=ulaw allow=alaw context = from-trunk ; Send unknown SIP callers to this context callerid = Unknown register=telno:passwd:[EMAIL PROTECTED] [username] allow=ilbcgsmspeexg726alaw ;currently only G728 and aLaw supported auth=md5 canreinvite=no context=from-pstn;seems to be ignored :-( disallow=all dtmfmode=auto fromdomain=sip1.voipbuster.com fromuser=username host=sip1.voipbuster.com nat=yes qualify=1000 realm=sip1.voipbuster.com secret=XXX type=friend username=username HTH! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
On Tue, January 31, 2006 10:43, Juergen K. Zick said: HI, all newer HFC-S cards will do. Depending on your application and system, you could easily ebaying an used Fritz!Card PCI or some active AVM B1 controller. Depending on the card you want to use you must se ZAPHFC or mIISDN/chan_isdn or chan_capi or mixtures with 2 different cards ... good luck, but there are enough HowTos available ... --Juergen For HFC-S cards you can also use vISDN!!! It supports TE and NT modes... It's still a bit immature (jitter and echo need work) but showing great promise! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT?: International number parsing
On Fri, January 27, 2006 23:47, Script Head said: What you're trying to accomplish can be easily done with an SQL query. You need to create a table of all the prefixes (international dial+country code+city/carrier) and join by that prefix. On 1/27/06, Damon Estep [EMAIL PROTECTED] wrote: Can anyone shed some light on rules that might make the task of parsing the country code and city codes from a dialed number in the CDRs? I know that there is almost never a case where a concatenated country and city code could overlap with another country code, but what about city codes and local numbers? Is it possible for a concatenated city code and local number to match another city code in the same country? I already have the table of country and city codes built. Are there holes in this theory; 1. Starting after the international dialing code, find the longest match for country code. 2. Starting after the country code from step 1, find the longest match for city code within that countries table of city codes. 3. The rest is the local number. Are there known exceptions? Am I reinventing the wheel rather than finding the right already existing resource? Obviously countrycodes are unique, and are created in a few 'classes' which also always provide unique numbers. Only one country has a single digit code: USA = 1 Most countries have a 2 digit code (31 = NL, 44 = UK, 49 = DE, etc.) There are *no* country codes with more than two digits that overlap the 2 digit codes. (So there's no 3 digit CC that starts with, for example, 31, 44, 49, etc.) So it is possible to 'categorize' them in to 1, 2, 3 digit CC's. Also the international dial codes have been chosen to not overlap anything else. So if you see (for instance) 011 you will always know it is an international call, and the next 1-3 digits will be a country code. -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AAH out bound routing problem
On Fri, January 27, 2006 15:13, ram said: Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700 -- Got SIP response 488 Not acceptable here back from (PeerIP) -- SIP/easycall-838e is circuit-busy ram Most likely the telno provided (19197543700) is not compatible with what they expect... Maybe you need to att digits (Perhaps 0019197543700) or remove digits? Or maybe you're not authenticated ? We'll need more info to be able to assist any further... To begin with it would help to know what configuration they expect... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External IAX2 phone defined as internal behaving as from PSTN
On Fri, January 27, 2006 16:09, Ian Cowley said: Have [EMAIL PROTECTED] 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring group macro. The same phone either on the same internal network to the asterisk or on a VPN to said network work fine. Obviously asterisk thinks this call is external. How do change this? SNIP The actual iax.conf part pertaining to this phone might be helpful here... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] External IAX2 phone defined as internal behaving as from PSTN
On Fri, January 27, 2006 17:23, Ian Cowley said: Iax.conf [general] ;bindport = 4569 ; Port to bind to (IAX is 4569) bindport = 5036 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=g729 ; 4 simultaneous allowed allow ilbc ; prefered for iax2 allow=gsm ; 13 Kbps (full rate), 20ms frame size allow=ulaw ;(g711)64 Kbps, sample-based allow=alaw ;(g711)64 Kbps, sample-based mailboxdetail=yes jitterbuffer=yes context=from-internal #include iax_additional.conf #include iax_custom.conf iax_additional.conf [1055] username=1055 type=friend secret=# record_out=Adhoc record_in=Adhoc qualify=yes port=4569 notransfer=yes [EMAIL PROTECTED] host=dynamic context=from-internal callerid=device 1055 Regards ianC Looks like you are using AMP / [EMAIL PROTECTED] As far as I can tell, this should work correctly... There might be something going on in the translation by the Checkpoint NAT control... Have you tried iax2 debug to see what it is receiving? the first few packets should give you sufficient information... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipbuster problem
On Tue, January 24, 2006 12:09, RumaTech said: Hi, all I have a problem using voipbuster (and voipstunt) for that matter. On all calls, voice is disconnected after 30s. Asterisk still thinks that call is in progress and I do not get any tones, just silience. Remote party gets normal tones for disconnection. I have paid my 10e, so it is not that. Technical support bever came back to me. I have used them before on IAX, now I am running SIP. Same here: IAX2 worked fine, SIP now works sometimes, partially and unreliably! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing the none commercial intel g729codecs into [EMAIL PROTECTED] 2.2?
On Sun, January 22, 2006 13:02, Charles Wang said: I have the same problem too. I install the G.729 (IPP) to asterisk 1.0.x, and it works well. When I change asterisk from 1.0.x to 1.2.x, and G.729 seems work fine. I can use show translation and find it too. But when I make a call using G.729. The asterisk (1.2.1) crashed. If i mark the line allow=g729 from sip.conf. And asterisk works fine. Just tested with 1.2 trunk to another 1.2 machine with g729, and all worked fine! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installing the none commercial intel g729codecs into [EMAIL PROTECTED] 2.2?
On Sun, January 22, 2006 19:40, Douglas Garstang said: Hang on there's a non commercial G729 codec that will work with Asterisk? Can someone point me to where I can find it? Thanks, Doug. Intel provides a sample for non-commercial/testing. http://www.voip-info.org/wiki-ITU+G.729 and http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+pass-thru The latter also has a link to the binaries... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is sip1.voipbuster.com corking reliably for others on list?
On Sun, January 22, 2006 22:32, Ron Wellsted said: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Guillermo Salas M wrote: I've the same problem with sip1.sipdiscount.com. The calls are not connecting but are billed. SIPDiscount seem to have been having intermittent problems since Friday morning. It seems to be working now however. Will be testing again tomorrow! ;-/ -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is sip1.voipbuster.com corking reliably for others on list?
I am trying to move from IAX2 to SIP for voipbuster, moving at the same time to sip1.voipbuster.com. When I try calling out, I see that there is SIP exchange, and in many cases also RTP data being exchanged. Hover in a very large number of attempts the connection is not established. Half of the time there is no RTP, the rest of the time there *is* RTP data flowing in two ways, but no ringtone is heard, and after a while the connection is terminated... Before I put in more time to investigate this, I should like to ask if people in general have any (good?) experience with VB's new SIP servers?... TIA BRgds -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2?
On Sat, January 21, 2006 22:10, MapsAir said: Has anyone successfully Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2? I tried to follow the instruction from http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ and http://aussievoip.com.au/tiki-index.php?page=G729-Install but I can't. I did it with [EMAIL PROTECTED] 1.5, but not 2.2 Working on it now... Will let you know how, if I succeed! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2?
On Sat, January 21, 2006 23:21, Franz Bräuer said: Hi, MapsAir wrote: Has anyone successfully Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2? Installed them today. Installing from source didn't work for me (Debian, Asterisk 1.2 from svn) but just adding the binaries (see the wiki on voip.org) did the job. Have you already tried the binaries? Kewl! Those work like a treat! As my testbox is a PII-750 running [EMAIL PROTECTED] 2.2 I did: cd /usr/lib/asterisk/modules/ wget http://kvin.lv/pub/Linux/Asterisk/codec_g723-gcc-pentium2.so wget http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium2.so After reloading, 'show translation' gives: Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 -22 8 817 8 724 115 19897 gsm 151 - 7 716 7 623 114 19796 ulaw 14616 - 111 2 118 109 19291 alaw 14616 1 -11 2 118 109 19291 g726 154241010 -10 926 117 20099 adpcm 14616 2 211 - 118 109 19291 slin 14515 1 110 1 -17 108 19190 lpc10 161311717261716 - 124 207 106 g729 16939252534252441 - 215 114 speex 16030161625161532 123 - 105 ilbc 17343292938292845 136 219 - Jolly good show, old chap! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AIX calls with sipdiscount
On Fri, January 20, 2006 21:46, Roberto Pereyra said: Hi Someone have luck using Sipdiscount service with IAX ? I only can use sipdiscount IAX service using a free account (only 1 minute call) , I have a normal account and with it can login in the IAX server. I using sip1.sipdiscount.com like IAX server but can make free calls (less 1 minute). Thanks in advance. roberto Finarea s.a. are discontinuing IAX, soon! So it's not worth the effort to try to make it work! Only iax.* / sip.* (same host) does IAX2. sip1.* is apparently an outsourced server which only supports SIP. conectionserver1.* is the server to which their own client connects. Not sure what exact protocols are involved there! HTH! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fritz card technology German *
On Thu, January 19, 2006 0:13, Hans Witvliet said: On Wed, 2006-01-18 at 11:45 +, John Daragon wrote: snip You can't use a Digium card because Digium doesn't make an ISDN2 card. snip If i see how many questions/complaints there are on the list about isdn/bri i would allmost wonder why digium does not make a single/quad active bri board Bri may not be popular as PRI in the usa, here in NL it's quite the opposite. PRI is way off limits for SOHO: it costs an arm and a leg initially and several toes a month ;-) I hear ya! We're using several BRI's rather than a PRI. We do not need the full complement of channels a PRI offers, but if prices were more reasonable we might have considered it anyway, simply because 1 PRI is much easier than several BRI's. Prices are so outrageous though that we settled for multiple BRI's and take the extra hassle for what it is... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fritz card technology German *
On Tue, January 17, 2006 22:10, Camilo Gonzalez-Cortes said: The Fritz cards was not designed to run on asterisk whereas the following German ISDN cards (http://www.junghanns.net/en/quadBRI_produkt.html) was designed specially to run on this platform. The only problem with this vendor is the support...It is terrible. They never respond an e-mail Almost any card with the cologne HFC-S chip will work with their drivers + Florz patch, mISDN or vISDN. In my epxerience vISDN gives the best EURO-ISDN support, but it is a very young project, and still misses crucial stuff like echo cancelling... It is moving at a high pace though, so keep an eye on it... BriStuff is the most mature, but also still has bugs, and contrary to the vISDN developer, they hardly ever respond to emails... Whatever you choose, good luck! :-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Test to see if I'm still on list...
As I haven't received any posts since yesterday... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: automon - one touch record
On Fri, January 13, 2006 8:51, Tomislav Parcina said: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Also: What are the SIP CanReinvite settings for these phones? This shuldn't be important because he have w and W in his dial plan. * doesn't allow reinvite if you have t, T, w or W. It shouldn't make a difference, but should not and does not isn't always the same thing! I like to be thorough and systematic when problem solving... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: automon - one touch record
On Fri, January 13, 2006 13:29, Tomislav Parcina said: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... It shouldn't make a difference, but should not and does not isn't always the same thing! We can't discus about this topic. It is simply meather of opinion. You think that is important and I don't. I like to be thorough and systematic when problem solving... Me to, that why I dont bother with erelevant things and care only about things that are relevant. Like I said before, it is mine and your opinion. It has no point discusing about it. In other words: Let's agree to disagree! ;-) That is fine with me... Have a nice weekend! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP and additional conf files
On Thu, January 12, 2006 19:18, Ben Ferguson said: Hello all. I've been searching and can't quite find what I'm looking for... I've gotten AMP installed and up and running quite decently on an Asterisk box and am now in the process of tweaking it to my needs. My company currently has around 70 employees and we are running on a complete Avaya system, but this system is no longer going to work for us (too much money for not enough stuff). So I have been put in charge of setting up an Asterisk PBX and get an entire test system going on it to see if Asterisk will meet our telephone needs. Extensions, queues, voicemail, stats, etc etc. Here's the problem: this Asterisk server is actually currently running live, serving information to people calling in to it. I need my test office setup, with AMP and this other system to work simultaneously, but yet totally separate. As my stuff is for a test, I would like to set it up so that when I dial in TO my Asterisk PBX FROM a specific telephone number, it takes me to my office test section in asterisk, otherwise, from ANY other number, it dials the info serving section. This would allow me to call from a certain telephone number and be able to get to my test office setup, but if anybody else calls from any other number, they get the other stuff. Doesn't sound too bad right? So how would one do this using AMP if AMP is more of the secondary system? If I understand correctly, to add additional, custom contexts to extensions.conf, it should be entered into extensions_additional.conf and the contexts should contain the word custom in them. So, first question, what if I want that custom context to be the first context (as in possibly the default context), but only if it's from a certain telephone number...? I assume you would enter that custom context as the context in zapata.conf, but how would you tell it to go back to the AMP stuff if the FROM telephone number is my speicifc telephone number? What context would I send it to so that it will do the regular AMP stuff? (Incidentally, I have a local telephone number and an 888 telephone number coming into my PRI, but when called, my Asterisk PBX views/receives them both as the local telephone number.) SNIP Normally in AMP (depending on version) you'd make either an inbound route like this : 4081234567|4081234599 (where the 4567 is the DID and 4599 the callerID) or an inbound route with DID=4081234567 and CID=4081234599 and then send it to a specific extension or custom context... HTH -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] automon - one touch record
On Fri, January 13, 2006 5:15, Jennifer Hales said: Hello all, I am unable to get automon recording to work; can someone advise me what I am doing wrong? When I do *1 all I see in the CLI screen is attempting native bridge of SIP/3006-291b and SIP/3153-6fdd, and there is no call record generated in /var/spool/asterisk/monitor/. Here are my settings: SNIP Does transferring with # or *2 work? (Or whatever sequences you assigned to those functions in feastures.conf...) That way you can get an idea whether it is just automon, or whether there's a more generic issue... Also: What are the SIP CanReinvite settings for these phones? Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer sounds - notifications
On Wed, January 11, 2006 12:46, Tomislav Parcina said: When I try to make attendend transfer (*2) this what hapends. I press *2 other person goes on hold and I hear transfer. I press extension number and that extension starts to ring but I don't hear anything. If nobody picks up that phone call in few seconds I get back to the person I was talking to (the person I triesd to transfer). The problem is that again, I don't hear anything (that person waits for me to say something) and I don't know that I'm back to transfered person. I hope that I have make it clear enough. Anyway, how can I solve this one? I would like to hear that the phone of extension is ringing, and I would like to konw when I'm speaking again with my caller. On http://www.voip-info.org/wiki-Asterisk+config+features.conf: ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call ;xfersound = beep ; to indicate an attended transfer is complete ;xferfailsound = beeperr; to indicate a failed transfer You could try these to see if that makes a difference?... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX CallerID
On Wed, January 11, 2006 7:52, scott said: Hi All Apologises if this has been disussed and I missed it. My SetUp I have a sip phone registered to an asterisk box (a1) in one location 1. This phone dials an extension which is in another location, so a1 passes the call via IAX to the other asterisk (a2) in location 2 which then dials the local phone. My Problem The caller ID setup in the sip.conf for the phone registered to a1 is not passed via the IAX to a2 and is therefor not being displayed on the phone in location2. The only way I can get the phone in location2 to display the caller ID is to set the callerid in the user part in the iax.conf on a2. Hope this makes sense Many thanks It sure does, as I am examining exactly the same issue for my set up... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX CallerID
On Wed, January 11, 2006 16:00, Colin Anderson said: As a rule of thumb, I always explicitly set CallerID in my dialplan before making a call through IAX, SIP or PSTN. If you make it part of a generic dialout routine then it isn't a hassle. It always works. It sometimes doesn't for my installation, but I'll check it later, it is not a big issue right now... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel modules load, but Asterisk fails at startup
On Wed, January 11, 2006 19:35, Stephen Bosch said: I'm running Asterisk on a Gentoo box with the Zaptel 1.2.1 drivers. If I boot the machine without having the wcfxs module autoload, then install the module with modprobe, asterisk works just fine. If I boot the machine and autoload the wcfxs module, the module loads fine: Jan 11 11:06:55 asterisk Zapata Telephony Interface Registered on major 196 Jan 11 11:06:55 asterisk ACPI: PCI Interrupt Link [LNKC] enabled at IRQ 10 Jan 11 11:06:55 asterisk PCI: setting IRQ 10 as level-triggered Jan 11 11:06:55 asterisk ACPI: PCI Interrupt :00:0a.0[A] - Link [LNKC] - GSI 10 (level, low) - IRQ 10 Jan 11 11:06:55 asterisk Freshmaker version: 73 Jan 11 11:06:55 asterisk Freshmaker passed register test Jan 11 11:06:55 asterisk Module 0: Installed -- AUTO FXS/DPO Jan 11 11:06:55 asterisk Module 1: Not installed Jan 11 11:06:55 asterisk Module 2: Not installed Jan 11 11:06:55 asterisk Module 3: Installed -- AUTO FXO (FCC mode) Jan 11 11:06:55 asterisk Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) The module is running: asterisk sfbosch # lsmod Module Size Used by wctdm 39936 - zaptel226756 - asterisk sfbosch # But Asterisk behaves as though it were not: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Jan 11 11:32:53 WARNING[5778]: chan_zap.c:920 zt_open: Unable to specify channel 1: No such device or address Jan 11 11:32:53 ERROR[5778]: chan_zap.c:6847 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jan 11 11:32:53 ERROR[5778]: chan_zap.c:10251 setup_zap: Unable to register channel '1' Jan 11 11:32:53 WARNING[5778]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Jan 11 11:32:53 WARNING[5778]: loader.c:554 load_modules: Loading module chan_zap.so failed! Warning, flexible rate not heavily tested! asterisk sfbosch # Ouch ... error while writing audio data: : Broken pipe Looking at this now as I write this, it seems that some module dependencies aren't loading, but I can't be sure. Does anybody have an idea what's going on here? -Stephen- Try running ztcfg -vvv -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel modules load, but Asterisk fails at startup
On Wed, January 11, 2006 21:36, Stephen Bosch said: Francesco Peeters (Asterisk) wrote: On Wed, January 11, 2006 19:35, Stephen Bosch said: Try running ztcfg -vvv Yes, that fixes it -- my question, I guess, is how to get that to run automatically at boot time... -s Either put it in rc.local or in /etc/modules or /etc/modprobe.conf or whatever the equivalent is on gentoo For example in my /etc/modprobe.conf: install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg alias wcfxs wctdm HTH -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel modules load, but Asterisk fails at startup
On Wed, January 11, 2006 23:37, Tzafrir Cohen said: On Wed, Jan 11, 2006 at 01:36:24PM -0700, Stephen Bosch wrote: Francesco Peeters (Asterisk) wrote: Try running ztcfg -vvv Yes, that fixes it -- my question, I guess, is how to get that to run automatically at boot time... I run ztcfg in a spcial init.d script for zaptel (which also does other clean-ups). Nothing stops you from running ztcfg in the asterisk init.d script. BTW: there is no point in the -vvv: ztcfg will be nice and verbose in reporting errors when they happen. No need for the extra noise (and wasted time) at boot. I agree about the -vvv being superfluous. I only added it to get confirmation that it actually had seen the card and it's ports in case it didn't work as expected... ;-) You may notice that there's no -vvv in the modprobe.conf sample lines either... Cheers! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Same Zap channel in multiple groups
On Mon, January 9, 2006 16:44, Patrick Conroy said: Does anyone know if it would cause problems to have the same Zap channel in multiple goups? So, for example, if I have two PRIs would the following work or would it cause problems: channel = 1-23 group = 1 channel = 25-47 group = 2 channel = 1-23,25-47 group = 3 I am just curious if anyone has set some thing like this up and how it worked out. Thanks, Patrick AFAIK group = 1,3 channel = 1-23 group = 2,3 channel = 25-47 should work... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Decent sub-$100 SIP phone.
On Tue, January 10, 2006 6:03, Dovid B. Asterisk Users said: Ken, I would tell the client that you offerd phones for under $100.00 and he didnt like them so now for a diffrent phone he will have to pay more. Also I have an 841 and for it works great. I also installed one for a customer in a mechanic shop and no complaints. Regards, Dovid I agree! They're the ones that don't want the 841. Also functionality is IMHO more important than looks, especially in an office/work environment. It'd be like getting a quote for a Suburban, then saying you don't like it and expecting an H2 for the same price instead... I would tell them that you'll need to requote for the phones... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommendations on a WiFi phone for *?
On Tue, January 10, 2006 5:50, Ira said: At 05:44 PM 01/09/2006, you wrote: We're getting our feet more and more wet with VOIP at work. We want to experiment with a good wireless (as in WiFi) phone. What would be a good phone to impress my boss with? I have the Zyxel P2000W V2 and while it has it's user interface annoyances, it's a great little phone and only $150 if you look hard enough. The most annoying one is sleeping, I guess to save battery life but if you forget to wake it up it looses the first 3 or 4 numbers you punch in. But it worked perfect, the first IP phone I've ever had and once I figured out I had to put the WEP code in hex it registered and work perfectly, even had people tell me how good I sound. Zyxel to an * box out a TDM400 to a Linksys VOIP router to ATT Callvantage. Ira Another, much cheaper option is to get DECT phones and connect them to IAXy's: DECT-PHONE ((( * ))) DECT-BASEIAXy[=IAX2=]Asterisk- TheWorld(tm) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording Calls at the phone
On Fri, January 6, 2006 15:37, Michael Sampson said: I work for a call center and we are looking at using asterisk to have our operators take calls. Our message taking software records all the calls on the operators computers. Right now we use these recording controls from radio shack that plug in between the wall jack and the phone and plug in via a 1/8 inch stereo connector to the mic input on the computer. If I buy an IP phone I can't do that. I could get an FXO adapter and regular phones, but I'm looking to get as little equipment as possible. Radio shack makes a recording control that plugs in to a 2.5 mm headset jack, but it takes batteries so thats not going to work Does anyone else do something similar? Does anyone have any ideas about what producs/setup would work for this. Asterisk has a built in monitoring system. You can chose to do Always, Never or On Demand monitoring, depending on your setup and dialplan http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcing a call transfer
On Fri, January 6, 2006 15:46, Michael Sampson said: With our current pbx system, a call comes in from the PSTN to the receptionist. She then hits flash, which puts the caller on hold, calls my extension, says so and so is on the phone for you, I say ok put him through, she hangs up and I am connected to the caller. With [EMAIL PROTECTED] I can it # then the extension to transfer to and it will ring there. But is there a simple way to announce the call before you transfer it. If not, does anyone have any good work arounds for this. -- It is called attended transfer. See http://www.voip-info.org/wiki/view/Asterisk+PBX+functions And http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf HTH! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Not Able to Connect Two Asterisk Servers Using IAX2
On Fri, January 6, 2006 20:20, Chandan Mishra said: Hi I have two asterisk servers. I just want to connect two asterisk server using IAX2. But the Asterisk Servers are not able to register each other. If some body have done this then Please send me the configuration they have done in iax.conf and extensions.conf. I simply want to connect and call from one sever to another. Thanks Chandan Kumar Mishra Software Engg. As always, the Wiki is your friend... http://www.voip-info.org/wiki-Asterisk+-+dual+servers I am using a modified version of method 3... You have to make sure that you have a user entry in IAX.conf for the other server as mentioned above... So if your serverA logs in using passwd SECRET, make sure that you have an entry [serverA] secret=SECRET type=user context=IncomingContext auth=md5(this one is optional of course...) Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
