Re: [Asterisk-Users] Caller ID with BT CD50
First of all thanks for the patch it works great, but i think it breaks the distinctive ringing, I have 2 incoming numbers in one x100p in contexts home1 and home2 but 'default' is always chosen has anyone else seen this? if you need any more info just ask Robb Tony Hoyle wrote: David J Carter wrote: Where would I find cdr-csv? Usually in /var/log/asterisk The line looks funny because of the line breaks. zapata.conf ukcallerid=yes callerid=asreceived signalling=fxs_ks channel = 1 : BT line channel = 2 : Telewest line I also have immediate=yes, but that shouldn't affect anything. Are you sure you've updated the modules correctly (done make/make install, done an rmmod on the old zaptel module and a modprobe on the new one)? There isn't much to go wrong beyond that... if you run asterisk with debugging you'll get a log if it finds a callerID but it's basically the same that goes into the cdr-csv file. Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to Asterisk - 2 question
Hi TH, Asterisk works fine as a Voicemail only server. I have it setup like that in a production setup. Configuration is simple, I will try and post something here soon. What will you integrate it with ? another asterisk system ? Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 28 May 2004 04:28 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to Asterisk - 2 question Hi All, I'm new to asterisk, and so far have yet to get past running the server up on a test PC. I have 2 Cisco 7960 phones to play with, both upgraded to the latest SIP image (7.1) I'd like to do 2 things, and hope that someone can point me to some simple documentation, example configs or other resources to get started: 1) A simple 1x1 setup, using the handsets described above, just to let me tinker and get an understanding of how Asterisk works. 2) A standalone voicemail server setup - Is it possible to use Asterisk just as a voicemail server ? If so, once again, any pointers to config examples etc would be appreciated. Thanks, TH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forwarding and record
Philipp von Klitzing wrote: Hi! my problem is to forwarding a call to a SIP phone and record the call at the same time. How can I do? This should help you to solve your problem: http://www.voip-info.org/wiki-Monitor+setup+sample Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, the problem is that the registration start before the answer of the forwarded call... Is it right? Best regards Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID with BT CD50
Cheers Tony. Your a star. Works a treat. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Hoyle Sent: 28 May 2004 00:48 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Caller ID with BT CD50 David J Carter wrote: Where would I find cdr-csv? Usually in /var/log/asterisk The line looks funny because of the line breaks. zapata.conf ukcallerid=yes callerid=asreceived signalling=fxs_ks channel = 1 : BT line channel = 2 : Telewest line I also have immediate=yes, but that shouldn't affect anything. Are you sure you've updated the modules correctly (done make/make install, done an rmmod on the old zaptel module and a modprobe on the new one)? There isn't much to go wrong beyond that... if you run asterisk with debugging you'll get a log if it finds a callerID but it's basically the same that goes into the cdr-csv file. Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: FireFly doesn't work with 3rd party anymore
I'm going to have to go against this statement, there's one bug that I need to fix so unfortunately it will have to be Monday now. For those after the IAX/SIP firefly (albeit an old version) get http://www.virbiage.com/firefly/download/firefly-dev.exe apologies, Adam Adam Hart wrote: They'll be a new version at the end of the day (it's 9:25am now) - The reason it was like that was to cope with overlap for the firefly network going to Freshtel. Freshtel will have the Firefly Network and special version of Firefly (no IAX and SIP) while Virbiage will have a standard IAX and SIP client. Freshtel has taken our Firefly Network to allow us to concentrate on Hardware (Insert vaporware joke here) If anyone's after Australian IAX termination (or Australians wishing to call overseas), try www.freshtel.net - iax server is ctsau.freshtel.net sorry for the dodgy version, Adam usedcanon wrote: Quite interesting, since there version history say 1.4 is the latest. The one you download is 1.7 and only works with Firefly. I have V1.5 which has the option to connect to other services. I am interested to know whats the highest version anyone has that has the other services options. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield Sent: 27 May 2004 19:30 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: FireFly doesn't work with 3rd party anymore In article [EMAIL PROTECTED], I wrote: In article [EMAIL PROTECTED], brian [EMAIL PROTECTED] wrote: Just an FYI FireFly no longer works with anything but the FireFly network. No more SIP, No more IAX. It was a damn good IAX client... too bad its crap now. Are you sure? http://www.virbiage.com/firefly/download/ still says the following: Standalone SIP / IAX mode: If you want to use Firefly on our Firefly phone network (with your own voicemail etc.) then you will need to register a phone number. However, you can also use Firefly as a SIP or IAX client on your own network. Well, I just downloaded the new 1.7 build from their website (from the same page that states the above), and I see what you mean. When I first ran the new version, it still used my old settings, and successfully connected to my Asterisk server. I looked in the Options dialog, and as you say, there is no third party option at all, only the option to connect to the Firefly network. Moreover, when I changed an unrelated option (sound output device), it then overwrote my settings in the registry with new settings for the Firefly network, Freshtel. Not impressed. Especially since in their FAQ they still explicitly say it can be used with Asterisk systems. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to Asterisk - 2 question
Umar, The plan is to integrate with a Cisco Callmanager. We currently have a very old VM system, based on a Netscape product that was installed before my time. The current project is to upgrade CM and replace the voicemail. I think Asterisk will do the job for us, now I just need to convince the boss. Any hints you can provide would be great. Thanks, Thomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of usedcanon Sent: Friday, 28 May 2004 5:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] New to Asterisk - 2 question Hi TH, Asterisk works fine as a Voicemail only server. I have it setup like that in a production setup. Configuration is simple, I will try and post something here soon. What will you integrate it with ? another asterisk system ? Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 28 May 2004 04:28 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to Asterisk - 2 question Hi All, I'm new to asterisk, and so far have yet to get past running the server up on a test PC. I have 2 Cisco 7960 phones to play with, both upgraded to the latest SIP image (7.1) I'd like to do 2 things, and hope that someone can point me to some simple documentation, example configs or other resources to get started: 1) A simple 1x1 setup, using the handsets described above, just to let me tinker and get an understanding of how Asterisk works. 2) A standalone voicemail server setup - Is it possible to use Asterisk just as a voicemail server ? If so, once again, any pointers to config examples etc would be appreciated. Thanks, TH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID with BT CD50
Robert Boardman [EMAIL PROTECTED] wrote: First of all thanks for the patch it works great, but i think it breaks the distinctive ringing, I have 2 incoming numbers in one x100p in contexts home1 and home2 but 'default' is always chosen has anyone else seen this? Yes - it does break the distinctive ring detection, but that's easily sorted out. The correct way would be to move the if (p-use_callerid == 2) code within the existing if (p-use_callerid) block, with a couple more if conditionals here and there. The quick way, however, is to apply the attached chan_zap.c hack over the top of Tony Hoyle's great work. In the standard chan_zap.c, you can't have distinctive ring detection unless you also need Caller*ID detection. My hack makes two changes: 1. Changes an else if into an if to get the world = USA Caller*ID code to run. This will waste a little time, but no more than we were wasting anyway, before Tony's patch was applied. 2. Comment out a line of code to ensure that we always answer after the first ring. We need the first ring to give the the distinctive ring code something to work with, of course. It works for me. Hopefully it'll work for you too. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ chan_zap.c.diff Description: Binary data
RE: dialogic was RE: [Asterisk-Users] Glare condition - How well does asteriskhandle?
Darren, yes, I'd be happy to help. I'll contact you off list to sort out the arrangements. I should warn you that it may be a wasted journey for you, as I really dont know if it will exhibit the problem. Tim. Storer, Darren [EMAIL PROTECTED] wrote: __ Hi Tim, TP So it _may_ not be a problem for me as NTL is a patchwork TP of smaller telcos, my area (Manchester) may be more up to TP date. TP Anyone know an easy way to tell what I've got ? TP (or will I have to ask NTL -gh) Pound to a penny you have ISDN 85. It's been reported via the list recently that only one NTL region in the UK has ISDN 110 (EuroISDN). If you are near Manchester and you're amenable, I'd like to ask if you'd mind me coming down to capture a trace of Asterisk failing with NTL's ISDN 85? (Pretty please etc.) I have a portable(ish) Asterisk server, with PRI, that I can bring along and the whole thing should take between 30 minutes and 1 hour to setup. The test can take place any time early or late (weekend's ok too) to suit you and the needs of your business. It would be great to move the ISDN 85 problem forward; I've lost access to the spare ISDN 85 circuit at a local switch site as it now has a production server on it... There are a number of features missing from ISDN 85 and some additional Information Elements that are sent, especially during call setup and tear down. I'm hoping that a patch to the existing Q.931 stack is all that's required but without some hard facts to go on it will be difficult to crack. Regards Darren -- Comgate TelcoInternetBroadcast Tel: +44(0)700 COMGATE -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of tim panton Sent: 27 May 2004 20:59 To: [EMAIL PROTECTED] Subject: Re: dialogic was RE: [Asterisk-Users] Glare condition - How well does asteriskhandle? Steve Underwood wrote: Jason Williams wrote: At 09:16 27/05/2004 -0500, you wrote: Maybe the time and effort would be better spent finding out why the Digium card won't work on the NTL's PRI and either fixing it or providing the information and testing facility to someone who can. NTL's PRI uses ISDN 85 not q931 so a ne protocol stack would need to be written. I think you means ISDN 85 not EuroISDN. Good heavens. I thought ISDN 85 died out in about 89. :-) I don't know where you would get the spec these days, but it shouldn't be a lot of work to modify libpri to add another variant of ISDN. I should say that I don't _know_ what NTL are delivering me, I haven't (yet) tried it with a digium E1 card. What I do know is : 1) the dialogic card claims to be running CTR4 on an E1 ISDN PRI 2) other folks on this list have had difficulty getting digium cards to talk to NTL. 3) exactly the same dialogic config works on BT and the Dutch PTT's E1 lines. So it _may_ not be a problem for me as NTL is a patchwork of smaller telcos, my area (Manchester) may be more up to date. Anyone know an easy way to tell what I've got ? (or will I have to ask NTL -gh) T. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: dialogic was RE: [Asterisk-Users] Glare condition - How well does asteriskhandle?
Hi Steve, SU If you are using CTR4, then I guess they use CTR4. :-) SU CTR4 == Net 5 == various other names == EuroISDN. Reasonable logic but bad assumption in this case. The Dialogic Q.931 stack (D/300, DM3 etc.) is solid and quite tolerant of ISDN 85 as are most hardware PBXs. Other (PC based) products exhibit exactly the same fussy behaviour though; the Digi RAS products ( http://tinyurl.com/36e7l ) work well with EuroISDN but won't work with ISDN 85 so the Asterisk stack is not alone in freaking when presented with this Frankenstein Protocol of the ISDN world. (Thanks a bunch BT/Marconi/GPT et al who rushed ISDN85 into service because they didn't want 18 months of effort to delay real Q.931 deployment in the UK, so they bolted a protocol converter on the end of existing DASS line cards instead of developing a native solution...ugly stuff!) I would like to try to help Tim decide which version of PRI he has as I'm local to him, let's see if he takes me up on the offer to plug a working * box into his PRI... Even if he has ISDN85 we would still benefit from the chance to capture the failure (using an MPA) and compare it to some good (working) * traces from a real EuroISDN circuit. Then the fun starts trying to find a neat way to patch the stack... Regards Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: 28 May 2004 01:39 To: [EMAIL PROTECTED] Subject: Re: dialogic was RE: [Asterisk-Users] Glare condition - How well does asteriskhandle? tim panton wrote: Steve Underwood wrote: Jason Williams wrote: At 09:16 27/05/2004 -0500, you wrote: Maybe the time and effort would be better spent finding out why the Digium card won't work on the NTL's PRI and either fixing it or providing the information and testing facility to someone who can. NTL's PRI uses ISDN 85 not q931 so a ne protocol stack would need to be written. I think you means ISDN 85 not EuroISDN. Good heavens. I thought ISDN 85 died out in about 89. :-) I don't know where you would get the spec these days, but it shouldn't be a lot of work to modify libpri to add another variant of ISDN. I should say that I don't _know_ what NTL are delivering me, I haven't (yet) tried it with a digium E1 card. What I do know is : 1) the dialogic card claims to be running CTR4 on an E1 ISDN PRI 2) other folks on this list have had difficulty getting digium cards to talk to NTL. 3) exactly the same dialogic config works on BT and the Dutch PTT's E1 lines. So it _may_ not be a problem for me as NTL is a patchwork of smaller telcos, my area (Manchester) may be more up to date. Anyone know an easy way to tell what I've got ? (or will I have to ask NTL -gh) T. If you are using CTR4, then I guess they use CTR4. :-) CTR4 == Net 5 == various other names == EuroISDN. It sounds like you are OK. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 Avm fritz passive card in the same box
Hi, I successfully installed 2 avm card
in my asterisk box but I'm unable to make call. My capi.conf is:
msn=072,0725
incomingmsn=*
controller=1,2
softdtmf=1
context=default
echocancel=yes
callgroup=1
devices=2,2
my capi info :
Contr1: 2 B channels total, 2 B channels
free.
Contr2: 2 B channels total, 2 B channels
free.
my extensions.conf :
exten = _0.,1,Dial(CAPI/072:b${EXTEN:1})
When I make a call I receive :
-- Executing Dial(SIP/2111-9940,
CAPI/072:b33511) in new stack
-- data ="">
-- capi request omsn =072
== found capi with omsn =072
May 28 10:36:56 NOTICE[180241]: app_dial.c:655
dial_exec: Unable to create channel of type 'CAPI'
== Everyone is busy at this time
Someone can help me ?
Re: [Asterisk-Users] Caller ID with BT CD50
Kevin, Could you add this to http://bugs.digium.com/bug_view_page.php?bug_id=0001719 Chris - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 28, 2004 9:12 AM Subject: RE: [Asterisk-Users] Caller ID with BT CD50 Robert Boardman [EMAIL PROTECTED] wrote: First of all thanks for the patch it works great, but i think it breaks the distinctive ringing, I have 2 incoming numbers in one x100p in contexts home1 and home2 but 'default' is always chosen has anyone else seen this? Yes - it does break the distinctive ring detection, but that's easily sorted out. The correct way would be to move the if (p-use_callerid == 2) code within the existing if (p-use_callerid) block, with a couple more if conditionals here and there. The quick way, however, is to apply the attached chan_zap.c hack over the top of Tony Hoyle's great work. In the standard chan_zap.c, you can't have distinctive ring detection unless you also need Caller*ID detection. My hack makes two changes: 1. Changes an else if into an if to get the world = USA Caller*ID code to run. This will waste a little time, but no more than we were wasting anyway, before Tony's patch was applied. 2. Comment out a line of code to ensure that we always answer after the first ring. We need the first ring to give the the distinctive ring code something to work with, of course. It works for me. Hopefully it'll work for you too. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk addons
Hi to all!! Is there another method to download asterisk addons??? Thanks F
[Asterisk-Users] Asterisk with Draytek 2600V
I am unable to get a my Draytek working with our Asterisk server. I can make/recieve calls but get no audio. I have tried the various codecs at the Vigor end but still getting nothing. I looked at sip debug (below) but am new to Asterisk and don't really know what I am looking for. Asterisk works fine with XLITE so I know my installation is ok. Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 From: phone1 sip:[EMAIL PROTECTED]:5060;tag=eSJ-4736 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED] Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp Content-Length: 290 v=0 o=phone2 5972727 56415 IN IP4 192.168.1.1 s=SIP Call c=IN IP4 192.168.1.1 t=0 0 m=audio 10116 RTP/AVP 18 0 8 4 2 101 a=rtpmap:18 G729/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:4 g723/8000 a=rtpmap:2 g726/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 13 lines Using latest request as basis request Sending to 192.168.1.1 : 5060 (non-NAT) Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 2 Found RTP audio format 101 Peer RTP is at port 192.168.1.1:0 Found description format G729 Found description format pcmu Found description format pcma Found description format g723 Found description format g726 Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x11d(G723|ULAW|ALAW|G726|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user 'phone1' Looking for 9080055 in sip list_route: hop: sip:[EMAIL PROTECTED] Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 From: phone1 sip:[EMAIL PROTECTED]:5060;tag=eSJ-4736 To: sip:[EMAIL PROTECTED];tag=as71701551 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.1.1:5060 We're at 192.168.0.250 port 13586 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 From: phone1 sip:[EMAIL PROTECTED]:5060;tag=eSJ-4736 To: sip:[EMAIL PROTECTED];tag=as71701551 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 265 v=0 o=root 24864 24864 IN IP4 192.168.0.250 s=session c=IN IP4 192.168.0.250 t=0 0 m=audio 13586 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 192.168.1.1:5060 mars*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-YQM-30118 From: phone1 sip:[EMAIL PROTECTED]:5060;tag=eSJ-4736 To: sip:[EMAIL PROTECTED];tag=as71701551 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Content-Length: 0 9 headers, 0 lines mars*CLI Sip read: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-eSd-21367 From: phone1 sip:[EMAIL PROTECTED]:5060;tag=eSJ-4736 To: sip:[EMAIL PROTECTED];tag=as71701551 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Content-Length: 0 9 headers, 0 lines Sending to 192.168.1.1 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-eSd-21367 From: phone1 sip:[EMAIL PROTECTED]:5060;tag=eSJ-4736 To: sip:[EMAIL PROTECTED];tag=as71701551 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.1.1:5060 Destroying call '[EMAIL PROTECTED]' mars*CLI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk console messages
was wondering if someone could give any indication of the messages that are appearing on the console of an Asterisk PBX WARNING[1116941120]: chan_sip.c:532 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (non-critical request) 192.168.90.1 is a 7940 ip phone configured as a SIP dial peer on asterisk pbx i mght added that the call seems to take place ok but this message appears every time - was hoping to some 'heads-up' on the severity of this message as it does seem to indicate some sort of failiure / misconfiguration ?? Thanks GT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 Avm fritz passive card in the same box
[EMAIL PROTECTED] wrote: msn=072,0725 [..] == found capi with omsn =072 May 28 10:36:56 NOTICE[180241]: app_dial.c:655 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time Are you sure, that your format for the msn definition is correct for Italy? In Germany we have to specify the local number only, no area code, no long distance access number, ie. having the following phone number 089-1234567 in Munich, would need msn=1234567 -- hth rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID with BT CD50
Just tried to apply the patch: Just checked out asterisk stable and zaptel, patched using Tony's patches (which worked, and compiled previously) Then got this when applying your patch. bash # cat ../chan_zap.c.diff | patch -p0 patching file channels/chan_zap.c Hunk #1 succeeded at 4642 (offset -148 lines). Hunk #2 FAILED at 4681. 1 out of 2 hunks FAILED -- saving rejects to file channels/chan_zap.c.rej bash # Cheers, Karl -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: 28 May 2004 09:12 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Caller ID with BT CD50 Robert Boardman [EMAIL PROTECTED] wrote: First of all thanks for the patch it works great, but i think it breaks the distinctive ringing, I have 2 incoming numbers in one x100p in contexts home1 and home2 but 'default' is always chosen has anyone else seen this? Yes - it does break the distinctive ring detection, but that's easily sorted out. The correct way would be to move the if (p-use_callerid == 2) code within the existing if (p-use_callerid) block, with a couple more if conditionals here and there. The quick way, however, is to apply the attached chan_zap.c hack over the top of Tony Hoyle's great work. In the standard chan_zap.c, you can't have distinctive ring detection unless you also need Caller*ID detection. My hack makes two changes: 1. Changes an else if into an if to get the world = USA Caller*ID code to run. This will waste a little time, but no more than we were wasting anyway, before Tony's patch was applied. 2. Comment out a line of code to ensure that we always answer after the first ring. We need the first ring to give the the distinctive ring code something to work with, of course. It works for me. Hopefully it'll work for you too. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Changes???
Hi Everybody Any significant changes to CVS HEAD over the last couple of days. I've got two asterisk boxes - both on public IP but one is dynamic. The one on dynamic IP registers at the other one - that part is fine. Calls going from the one with dynamic to the static one goes fine. Call the other way results now in: Failed to authenticate user 1101 sip:[EMAIL PROTECTED] 1101 is a SIP phone authenticated at the static server. All sip entries have canreinvite=no. Two days ago this was working fine. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Pascal
usedcanon [EMAIL PROTECTED] wrote: Thanks, suddenly makes sense now. I guessed that is the case however was not sure. Any opinion on what is more/most efficient, using a scripting language like perl or a compile app in C/pascal. Define efficient. A C program would normally be expected to be about ten times faster than a Perl script. But when it's 10ms to execute instead of 100ms, it probably doesn't matter. If your time is not free, it may be more efficient to write a quick script in Perl and buy a faster server than it is to spend ages writing in C. Either way, if you're spending anything bit a trivial amount of CPU time executing AGI scripts (whatever the language), you've probably misdesigned something. So the ultimate answer is that AGI scripts should be written in whatever language you're most comfortable doing them in. -- Vice is its own reward. It is virtue which, if it is to be marketed with consumer appeal, must carry Green Shield stamps. - Quentin Crisp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Pascal
hi Peter, Your feedback is greatly appreciated. Having not done any AGI before I was not sure what to expect. My requirements are very basic at the moment, and time as you say is money. my best option is to find something simmillar and customise it to my needs. Umar. --- Peter Corlett [EMAIL PROTECTED] wrote: usedcanon [EMAIL PROTECTED] wrote: Thanks, suddenly makes sense now. I guessed that is the case however was not sure. Any opinion on what is more/most efficient, using a scripting language like perl or a compile app in C/pascal. Define efficient. A C program would normally be expected to be about ten times faster than a Perl script. But when it's 10ms to execute instead of 100ms, it probably doesn't matter. If your time is not free, it may be more efficient to write a quick script in Perl and buy a faster server than it is to spend ages writing in C. Either way, if you're spending anything bit a trivial amount of CPU time executing AGI scripts (whatever the language), you've probably misdesigned something. So the ultimate answer is that AGI scripts should be written in whatever language you're most comfortable doing them in. -- Vice is its own reward. It is virtue which, if it is to be marketed with consumer appeal, must carry Green Shield stamps. - Quentin Crisp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger - Communicate instantly...Ping your friends today! Download Messenger Now http://uk.messenger.yahoo.com/download/index.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] generate dial tone
On May 27, 2004, at 11:01 PM, Aaron J. Angel wrote: Michael George wrote: But, this isn't a big deal, we can live without it. I just thought there might be a way. If I could do a Backtround(Playtone()), that would do what I want... There's no need for that. The playtone application continues to the next priority as it plays the tone, and keeps playing the tone until you call stoptone[s?]. Playtone(dial) should be what you're looking for then, and at the extension t (if you want a timeout), just call stoptone (or stoptones, I don't remember). Check out www.voip-info.org. Yes, I see what you are saying. And I tried this. Here's what happens: I get the 9 and start PlayTones(). I go to the next context (with the tones playing). In the next context (tones still playing) my matches are all several digits long, so the tone is playing as the digits are pressed. That is disorienting because that usually happens on a broken line. However, if you notice how Background() works, it will play the sound file and still accept input. Once it gets the first input key it will stop playing and begin its matching. That is exactly the behavior I want. Now, I thought I could do playtones() and then match the just first input number (0, 1, or N). On 0, 1 or N (in separate extensions, of course), I would stopplaytones() and then goto() the next context (international, long distance, local -- respectively). The int and ld contexts are straightforward, but the new local context needs to know which extension was dialed (the 'N') to complete the calling. I tried that yesterday and got frustrated at the resulting complexity of trying to do such a simple and inconsequential thing. I figured that the cost outweighed the benefit and I need to get this prototype going so that we can move into full launch. This dialtone issue needs to become a tier 2 or tier 3 feature. -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID with BT CD50
Karl Dyson [EMAIL PROTECTED] wrote: Just checked out asterisk stable and zaptel, patched using Tony's patches (which worked, and compiled previously) Then got this when applying your patch. bash # cat ../chan_zap.c.diff | patch -p0 patching file channels/chan_zap.c Hunk #1 succeeded at 4642 (offset -148 lines). Hunk #2 FAILED at 4681. 1 out of 2 hunks FAILED -- saving rejects to file channels/chan_zap.c.rej bash # Could you try applying the changes by hand. There are only two lines to change and it looks as if the first one went through. I'll check my patch to see if I messed up the original or something silly. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
Karl Dyson wrote: Just tried to apply the patch: Just checked out asterisk stable and zaptel, patched using Tony's patches (which worked, and compiled previously) Then got this when applying your patch. bash # cat ../chan_zap.c.diff | patch -p0 patching file channels/chan_zap.c Hunk #1 succeeded at 4642 (offset -148 lines). Hunk #2 FAILED at 4681. 1 out of 2 hunks FAILED -- saving rejects to file channels/chan_zap.c.rej bash # The patch is against the HEAD branch not the stable one. Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 Phone(FWD): (0845 004 5566) 413300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] generate dial tone
Michael George [EMAIL PROTECTED] wrote: I get the 9 and start PlayTones(). I go to the next context (with the tones playing). In the next context (tones still playing) my matches are all several digits long, so the tone is playing as the digits are pressed. That is disorienting because that usually happens on a broken line. However, if you notice how Background() works, it will play the sound file and still accept input. Once it gets the first input key it will stop playing and begin its matching. That is exactly the behavior I want. Now, I thought I could do playtones() and then match the just first input number (0, 1, or N). On 0, 1 or N (in separate extensions, of course), I would stopplaytones() and then goto() the next context (international, long distance, local -- respectively). The int and ld contexts are straightforward, but the new local context needs to know which extension was dialed (the 'N') to complete the calling. I tried that yesterday and got frustrated at the resulting complexity of trying to do such a simple and inconsequential thing. I figured that the cost outweighed the benefit and I need to get this prototype going so that we can move into full launch. This dialtone issue needs to become a tier 2 or tier 3 feature. Have you not looked at the DISA application (command) yet? That seems to me to be a much better solution to your problem. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cvs problem with TDM04B ?
I there a problem with CVS ? My card TDM04B does not want to answer calls on 2 ports. Strange. Yes there is a problem. Pull an older copy of wcfxs.c in zaptel (from about 5/24) and it will work again. Mark is aware of the problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID with BT CD50
Oddly, it looks like the changes were made(!?) It might be, having read Tony's reply, that it's because I applied the uk cli patches from Tony and yourself to the stable rather than head branches? I'll try compiling and let you know. Cheers for now, Karl -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: 28 May 2004 12:07 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Caller ID with BT CD50 Karl Dyson [EMAIL PROTECTED] wrote: Just checked out asterisk stable and zaptel, patched using Tony's patches (which worked, and compiled previously) Then got this when applying your patch. bash # cat ../chan_zap.c.diff | patch -p0 patching file channels/chan_zap.c Hunk #1 succeeded at 4642 (offset -148 lines). Hunk #2 FAILED at 4681. 1 out of 2 hunks FAILED -- saving rejects to file channels/chan_zap.c.rej bash # Could you try applying the changes by hand. There are only two lines to change and it looks as if the first one went through. I'll check my patch to see if I messed up the original or something silly. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call forwarding
I am using CISCO 30 VIP and CP 12+ IP phones. I am using 2 analog phones connected to a SIPURA. I am using chan_skinny for the CISCO phones. On the CISCO phones, only the basic phone functionality works. I can not transfer calls or anything using the chan_skinny. The analog phones also work as basic phones. From my earlier emails, I found out that chan_skinny does not support the advanced feature like this. Chan_sccp did not work with these two types of CISCO phones. I am looking for at least one phone in the system which can be the operator phone. I expect this phone to receive calls and if necessary transfer the call to an extension. Is there any possibility that I can do that with my existing phones. Otherwise, which are the recommended phones to get this functionality? Thanks, Naren __ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Changes???
Hi Lars, I met the same problems yesterday and even posted it to the list. Unfortunately nobody answered yet. Is it so clear to solve that no one is willing to help us? :-/ Regards, Julian Pawlowski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID with BT CD50
Well compiles and runs OK, but it doesn't identify the dring. I only started playing with it this morning (only realised it *did* dring when I saw your it's broken dring post) This is what I have in zapata.conf dring1=95,0,0 dring1context=inbound-pstn-1 dring2=325,95,0 dring2context=inbound-pstn-2 is this correct for the UK? (I suspect not, and yes, I have dring on my bt line). Cheers, Karl -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Karl Dyson Sent: 28 May 2004 12:36 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Caller ID with BT CD50 Oddly, it looks like the changes were made(!?) It might be, having read Tony's reply, that it's because I applied the uk cli patches from Tony and yourself to the stable rather than head branches? I'll try compiling and let you know. Cheers for now, Karl -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: 28 May 2004 12:07 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Caller ID with BT CD50 Karl Dyson [EMAIL PROTECTED] wrote: Just checked out asterisk stable and zaptel, patched using Tony's patches (which worked, and compiled previously) Then got this when applying your patch. bash # cat ../chan_zap.c.diff | patch -p0 patching file channels/chan_zap.c Hunk #1 succeeded at 4642 (offset -148 lines). Hunk #2 FAILED at 4681. 1 out of 2 hunks FAILED -- saving rejects to file channels/chan_zap.c.rej bash # Could you try applying the changes by hand. There are only two lines to change and it looks as if the first one went through. I'll check my patch to see if I messed up the original or something silly. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfering
I am using CISCO 30 VIP and CP 12+ IP phones. I am using 2 analog phones connected to a SIPURA. I am using chan_skinny for the CISCO phones. On the CISCO phones, only the basic phone functionality works. I can not transfer calls or anything using the chan_skinny. The analog phones also work as basic phones. From my earlier emails, I found out that chan_skinny does not support the advanced feature like this. Chan_sccp did not work with these two types of CISCO phones. I am looking for at least one phone in the system which can be the operator phone. I expect this phone to receive calls and if necessary transfer the call to an extension. Is there any possibility that I can do that with my existing phones. Otherwise, which are the recommended phones to get this functionality? Thanks, Naren __ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] generate dial tone
I did take a quick look at it, but the header indicated that DISA allows incoming calls to dial back out. I am just trying to emulate the feel of our current PBX which will just connect us to an outgoing line (with a dialtone) when we hit 9. (Though I don't want asterisk to mimic that behavior because I want to be judicious about which outgoing channels are used depending on the number dialed.) Am I mistaken on the use of DISA? On May 28, 2004, at 7:11 AM, Kevin Walsh wrote: Michael George [EMAIL PROTECTED] wrote: I get the 9 and start PlayTones(). I go to the next context (with the tones playing). In the next context (tones still playing) my matches are all several digits long, so the tone is playing as the digits are pressed. That is disorienting because that usually happens on a broken line. However, if you notice how Background() works, it will play the sound file and still accept input. Once it gets the first input key it will stop playing and begin its matching. That is exactly the behavior I want. Now, I thought I could do playtones() and then match the just first input number (0, 1, or N). On 0, 1 or N (in separate extensions, of course), I would stopplaytones() and then goto() the next context (international, long distance, local -- respectively). The int and ld contexts are straightforward, but the new local context needs to know which extension was dialed (the 'N') to complete the calling. I tried that yesterday and got frustrated at the resulting complexity of trying to do such a simple and inconsequential thing. I figured that the cost outweighed the benefit and I need to get this prototype going so that we can move into full launch. This dialtone issue needs to become a tier 2 or tier 3 feature. Have you not looked at the DISA application (command) yet? That seems to me to be a much better solution to your problem. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] JTAPI Interface in Asterisk
Title: Message Is there an interface (direct or indirect)in Asterisk that can be used by JTAPI to do third party call control and the other functionality supported by JTAPI? Does anyone have an example of such a thing? Jim
RE: [Asterisk-Users] Caller ID with BT CD50
Karl Dyson [EMAIL PROTECTED] wrote: Well compiles and runs OK, but it doesn't identify the dring. I only started playing with it this morning (only realised it *did* dring when I saw your it's broken dring post) This is what I have in zapata.conf dring1=95,0,0 dring1context=inbound-pstn-1 dring2=325,95,0 dring2context=inbound-pstn-2 is this correct for the UK? (I suspect not, and yes, I have dring on my bt line). I have this on my home setup: dring1 = 367,0,0 dring1context = incoming-pstn-personal dring2 = 247,0,0 dring2context = incoming-pstn-business -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] INTERTEX AND ASTERISK
Hi all, I just upgrade my ix66 ... the new firmware 2.07 have this: (SIP) Tolerance against Asterisk PBX registration deviation. regards Miklos
Re: [Asterisk-Users] Caller ID with BT CD50
Kevin Walsh wrote:
Yes - it does break the distinctive ring detection, but that's easily
sorted out.
Actually it's the first time I've ever heard of distinctive ring being
available in the UK... :)
The correct way would be to move the if (p-use_callerid == 2)
code within the existing if (p-use_callerid) block, with a couple
more if conditionals here and there. The quick way, however, is
to apply the attached chan_zap.c hack over the top of Tony Hoyle's
great work.
It needs an extra conditional - no need to add the extra delay before
the phone starts ringing if there's no need to (I prefer my phone to
start ringing immediately).
Try this patch. It also enables distinctive ring detection even if
usecallerid=no. It's not well tested yet (well, at all actually since I
don't have access to distinctive ring...)
Tony
--
All your code belongs to Santa
Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
Phone(FWD): (0845 004 5566) 413300
? channels/chan_zap.cx
Index: callerid.c
===
RCS file: /usr/cvsroot/asterisk/callerid.c,v
retrieving revision 1.16
diff -u -r1.16 callerid.c
--- callerid.c 4 May 2004 06:42:06 - 1.16
+++ callerid.c 25 May 2004 20:04:27 -
@@ -134,6 +134,12 @@
return cid;
}
+void callerid_set_v23(struct callerid_state *cid)
+{
+ cid-fskd.f_mark_idx = 4; /* 1300 Hz */
+ cid-fskd.f_space_idx = 5; /* 2100 Hz */
+}
+
void callerid_get(struct callerid_state *cid, char **name, char **number, int *flags)
{
*flags = cid-flags;
@@ -255,7 +260,7 @@
break;
case 2: /* Number */
case 3: /* Number (for Zebble) */
- case 4: /* Number */
+ case 4: /* Number (UK: Reason for
number withheld) */
res = cid-rawdata[x];
if (res 32) {
ast_log(LOG_NOTICE,
Truncating long caller ID number from %d bytes to 32\n, cid-rawdata[x]);
@@ -266,7 +271,7 @@
cid-number[res] = '\0';
break;
case 7: /* Name */
- case 8: /* Name */
+ case 8: /* Name (UK: Reason for
absence of name) */
res = cid-rawdata[x];
if (res 32) {
ast_log(LOG_NOTICE,
Truncating long caller ID name from %d bytes to 32\n, cid-rawdata[x]);
@@ -275,6 +280,11 @@
memcpy(cid-name, cid-rawdata
+ x + 1, res);
cid-name[res] = '\0';
break;
+ case 17: /* Call type (UK) */
+ /* Currently defined: 1 =
Voice call, 2 = Ringback when free, 129 = Message waiting */
+ break;
+ case 19: /* Network message system
status (UK) */
+ break;
case 22: /* Something French */
break;
default:
Index: coef_in.h
===
RCS file: /usr/cvsroot/asterisk/coef_in.h,v
retrieving revision 1.1
diff -u -r1.1 coef_in.h
--- coef_in.h 20 Mar 2001 20:11:26 - 1.1
+++ coef_in.h 25 May 2004 16:55:41 -
@@ -6,4 +6,8 @@
}, {
9.8539686961e-02,-5.6297236492e-02,4.2915323820e-01,-1.2609358633e+00,2.2399213250e+00,-2.9928879142e+00,2.5990173742e+00,0.00e+00,
}, }, { {
1.8229206610e-04,-7.8997325866e-01,-7.7191410839e-01,-2.8075643964e+00,-1.6948618347e+00,-3.0367273700e+00,-9.0333559408e-01,0.00e+00,
}, {
9.8531161839e-02,-5.6297236492e-02,-1.1421579050e-01,-4.8122536483e-01,-4.0121072432e-01,-7.4834487567e-01,-6.9170822332e-01,0.00e+00,
- }, },
+ }, }, { { 1.8229206611e-04,-7.8997325866e-01, 2.5782298908e+00,
-5.3629717478e+00, 6.5890882172e+00, -5.8012914776e+00, 3.0171839130e+00,
-0.00e+00,
+ }, { 9.8534230718e-02,-5.6297236492e-02, 3.8148618075e-01, -1.0848760410e+00,
Re: [Asterisk-Users] Caller ID with BT CD50
Karl Dyson wrote: Well compiles and runs OK, but it doesn't identify the dring. I only started playing with it this morning (only realised it *did* dring when I saw your it's broken dring post) This is what I have in zapata.conf dring1=95,0,0 dring1context=inbound-pstn-1 dring2=325,95,0 dring2context=inbound-pstn-2 is this correct for the UK? (I suspect not, and yes, I have dring on my bt line). I expect you'll need usedistinctiveringdetection=yes as well. To my untrained eye it looks like the patch aborts after the first ring... I've done it in a slightly different way which may (or may not) work better. Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 Phone(FWD): (0845 004 5566) 413300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk addons
In article [EMAIL PROTECTED], Fabio Donaggio [EMAIL PROTECTED] wrote: Hi to all!! Is there another method to download asterisk addons??? Another method in addition to what? Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] generate dial tone
Michael George [EMAIL PROTECTED] wrote: I did take a quick look at it, but the header indicated that DISA allows incoming calls to dial back out. I am just trying to emulate the feel of our current PBX which will just connect us to an outgoing line (with a dialtone) when we hit 9. (Though I don't want asterisk to mimic that behavior because I want to be judicious about which outgoing channels are used depending on the number dialed.) I can't say that I fully understand the difference between the two cases you outlined above, but if DISA is not right for you then that's fine; That would probably explain why everyone else was suggesting ignorepat and the like. Am I mistaken on the use of DISA? If you call DISA(no-password,your-context-name) then it'll present a dial tone (with no password prompted for) and allow the user to dial numbers accessible from the specified context. Obviously, you'd have to be very careful to not allow anonymous incoming callers to dial '9' and then dial anything they like. I don't know how you plan to handle that case. DISA will prompt the user for a PIN unless you use the no-password keyword, so that'll go some way toward the security of your system. The named context, passed as an argument to DISA, would seem to satisfy your I want to be judicious about which outgoing channels are used depending on the number dialled requirement. All I can suggest is that you try it on a closed system and see if it does what you need. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Changes???
Julian Pawlowski [EMAIL PROTECTED] wrote: I met the same problems yesterday and even posted it to the list. Unfortunately nobody answered yet. Is it so clear to solve that no one is willing to help us? :-/ It sometimes helps if you quote some context above your text. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] generate dial tone
On Fri, 28 May 2004, Michael George wrote: Yes, I see what you are saying. And I tried this. Here's what happens: I get the 9 and start PlayTones(). I go to the next context (with the tones playing). In the next context (tones still playing) my matches are all several digits long, so the tone is playing as the digits are pressed. That is disorienting because that usually happens on a broken line. However, if you notice how Background() works, it will play the sound file and still accept input. Once it gets the first input key it will stop playing and begin its matching. That is exactly the behavior I want. http://bugs.digium.com/bug_view_page.php?bug_id=745 Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Downgrading Asterisk
The code changes that fixed the cisco choppy sound for Stable went in last Friday. That change corrected iax2 issues that had been known for well over a month but never got applied to Stable. That same code is in Head, however many other changes have happened to Head, and some of those apparently have impacted at least some of us (mostly cisco users). Stable has a number of other bugs that reportedly will never get fixed as the fixes use functionality that exists only in Head. It seems the choppy (and almost unusable) audio in Head is only impacting some cisco users, and since these problems are not impacting the few that can read code, use cisco phones, and are impacted, we're stuck with the problem. The problem seems to be very evasive, however switching the iax2 links to use only iLBC (and not gsm) has corrected issues for some. Although many of us that have worked in a production I/T arena assume something called Stable would truly have known bugs fixed, that's hardly the case for *. That branch really should be renamed to something like v1.0 and remove any reference to Stable and bug fixes as its treated as a lockdown for added functionality, and has nothing to do with functional stability. FYI Downgrading to -stable totally fixed the choppy audio on Cisco my 7960 - * - IAX setup. Now, when would a fix that goes into stable get into the current source (HEAD)? And, isn't checking stuff into a stable branch that doesn't exist elsewhere in the source tree break some rules somewhere? It has to. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nik Martin Sent: Tuesday, May 25, 2004 2:53 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Downgrading Asterisk I upgraded to the latest HEAD version of asterisk, and all IAX calls started sounding choppy. It was suggested on the IRC channel that I go back to asterisk -stable to determine if that fixes it. Is downgrading as simple as upgrading? Because now, -stable builds fine, but I get an error on the asterisk console when starting, something about ast_get_txt not found. Recompiling and installing asterisk HEAD afterwards works just fine. As a side note, I recently upgraded my kernel to 2.4.26 and had an issue with old kernel headers, but have since resolved that prior to trying this downgrade. Any ideas? Nik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Sound Card and No Sound from Phone
Hi! Newbie question; my server has no sound card, in effect I have commented out the loading of alsa and oss modules. When I make a call I do not here any sound however I do notice the activity from the tethereal trace and the debug. Is there a relation? I would think so but I am just shocked I have not noticed it mentioned anywhere, does why I am here now! -- Nana Yaw -- An Asterisk fun :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID with BT CD50
Tony Hoyle [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Yes - it does break the distinctive ring detection, but that's easily sorted out. Actually it's the first time I've ever heard of distinctive ring being available in the UK... :) It costs the same as Caller*ID, so I just got it to separate business and personal calls received at home. I have Asterisk now, so I can be a little bit cleverer and set the times when I want to receive business calls at home etc. :-) The correct way would be to move the if (p-use_callerid == 2) code within the existing if (p-use_callerid) block, with a couple more if conditionals here and there. The quick way, however, is to apply the attached chan_zap.c hack over the top of Tony Hoyle's great work. It needs an extra conditional - no need to add the extra delay before the phone starts ringing if there's no need to (I prefer my phone to start ringing immediately). Yes, but it'd need to ring at least once if the distinctive ring is to be detected. As you said - a conditional should do the trick. Try this patch. It also enables distinctive ring detection even if usecallerid=no. It's not well tested yet (well, at all actually since I don't have access to distinctive ring...) I applied your new patch but it resulted in the caller hearing a ring tone but no phones actually ringing. I don't have time to look into it right now, but I'll take a look later and see what's going on. I put my chan_zap.c back in and the re-tests were ok. I has assumed that the only change between your previous patch file and your latest was in chan_zap.c, so I extracted the patch and applied it to that single file only. If that's not the case then I'll have to apply the whole thing. I'll probably have more time later in the day. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk addons
- Original Message - From: Fabio Donaggio To: [EMAIL PROTECTED] Sent: Friday, May 28, 2004 6:16 AM Subject: [Asterisk-Users] Asterisk addons Hi to all!! Is there another method to download asterisk addons??? Thanks F Man! Try to investigate for yourself! Use google! http://www.google.com/search?q=asterisk-addons+downloadie=UTF-8hl=esmeta = Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and MySQL
Hi to all!! I'm successful to connect Asterisk to MySQL database... Can anyone learn me how to store sip user in MySQL database and how to configure voicemail?? Thanks for all!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Downgrading Asterisk
On Fri, 2004-05-28 at 07:59, Rich Adamson wrote: Although many of us that have worked in a production I/T arena assume something called Stable would truly have known bugs fixed, that's hardly the case for *. That branch really should be renamed to something like v1.0 and remove any reference to Stable and bug fixes as its treated as a lockdown for added functionality, and has nothing to do with functional stability. This comment shows you suffer from not understanding that words have more than one meaning. Stable means not changing much. A stable table doesn't fall over and not that it doesn't have flaws in the design such as being only 1 foot off of the ground. Similar people have the same mistaken opinion about Debian, it is stable because it doesn't change much. Only things that must change(security) gets changed in stable. Someone who runs stable shouldn't have to worry too much about things changing. Remember the reason for stable, it is there to make a run at a 1.0 code release. What software do you know of besides Hello World has a bug free 1.0 release. Please watch the inflammatory tone of your message next time you criticize the free software you are using and the people giving you their time. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS login
Hermann Wecke wrote: On Thu, 27 May 2004, Harry Flink wrote: www.cvshome.org is home for CVS but the site is currently down. Is down due to security issues: I'm surprised that was exploitable... it's much more likely to crash the server than do anything nasty. That's the patch that sourceforge used that broke the date handling I see... I'd have thought they would have come up with a better one by now - many sites will be unable to apply it because it renders many clients incompatible. Anyway, this is OT for this list :) Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 Phone(FWD): (0845 004 5566) 413300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] generate dial tone
It's true, if you're not careful, you could give incoming callers access to your outside lines. But it is possible, with careful use of contexts, to ensure that callers coming in on the context you specify for incoming calls does not have access to the context that contains the dialplan for outside calling. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Fri, 28 May 2004, Michael George wrote: I did take a quick look at it, but the header indicated that DISA allows incoming calls to dial back out. I am just trying to emulate the feel of our current PBX which will just connect us to an outgoing line (with a dialtone) when we hit 9. (Though I don't want asterisk to mimic that behavior because I want to be judicious about which outgoing channels are used depending on the number dialed.) Am I mistaken on the use of DISA? On May 28, 2004, at 7:11 AM, Kevin Walsh wrote: Michael George [EMAIL PROTECTED] wrote: I get the 9 and start PlayTones(). I go to the next context (with the tones playing). In the next context (tones still playing) my matches are all several digits long, so the tone is playing as the digits are pressed. That is disorienting because that usually happens on a broken line. However, if you notice how Background() works, it will play the sound file and still accept input. Once it gets the first input key it will stop playing and begin its matching. That is exactly the behavior I want. Now, I thought I could do playtones() and then match the just first input number (0, 1, or N). On 0, 1 or N (in separate extensions, of course), I would stopplaytones() and then goto() the next context (international, long distance, local -- respectively). The int and ld contexts are straightforward, but the new local context needs to know which extension was dialed (the 'N') to complete the calling. I tried that yesterday and got frustrated at the resulting complexity of trying to do such a simple and inconsequential thing. I figured that the cost outweighed the benefit and I need to get this prototype going so that we can move into full launch. This dialtone issue needs to become a tier 2 or tier 3 feature. Have you not looked at the DISA application (command) yet? That seems to me to be a much better solution to your problem. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and MySQL
On Fri, 2004-05-28 at 08:13, Fabio Donaggio wrote: Hi to all!! I'm successful to connect Asterisk to MySQL database... Can anyone learn me how to store sip user in MySQL database and how to configure voicemail?? Can I learn ya with a 2x4? BTW, what happened to your postgres connection you told us about yesterday? Also did you install CVS yet? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk console messages
Graham Turner wrote: was wondering if someone could give any indication of the messages that are appearing on the console of an Asterisk PBX WARNING[1116941120]: chan_sip.c:532 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (non-critical request) 192.168.90.1 is a 7940 ip phone configured as a SIP dial peer on asterisk pbx i mght added that the call seems to take place ok but this message appears every time - was hoping to some 'heads-up' on the severity of this message as it does seem to indicate some sort of failiure / misconfiguration ?? Without a SIP DEBUG or SIP history I can't say what message it was that failed, but it says non-critical, so it can be an OPTIONS or a NOTIFY. Turn on SIP debugging and you'll see thte message that is being retransmitted until cancelled. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
Kevin Walsh wrote: I applied your new patch but it resulted in the caller hearing a ring tone but no phones actually ringing. I don't have time to look into it right now, but I'll take a look later and see what's going on. I put my chan_zap.c back in and the re-tests were ok. I've changed it around slightly as (I think) the last one will result in the phone getting the ringtone noise when they pick up.. it was bypassing the actual reading of the data, whereas it's better to read and ignore it. I put that one on the web page. Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 Phone(FWD): (0845 004 5566) 413300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Changes???
Lars Boegild Thomsen wrote: Hi Everybody Any significant changes to CVS HEAD over the last couple of days. I've got two asterisk boxes - both on public IP but one is dynamic. The one on dynamic IP registers at the other one - that part is fine. Calls going from the one with dynamic to the static one goes fine. Call the other way results now in: Failed to authenticate user 1101 sip:[EMAIL PROTECTED] At which server? 1101 is a SIP phone authenticated at the static server. All sip entries have canreinvite=no. Two days ago this was working fine. Yes, there's been quite a lot of changes to SIP registration and authentication. So SIP calls from a user reigstred at the static server to an extension on the dynamic server doesn't work? Is this the setup: SIP phone 1101 - SIP CALL - Static server - SIP CALL - dynamic server ´..and the dynamic server is registred with the static server? Please add a SIP debug of the call so we can see what happens, who refuses what call. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * as pri_net?
If you have used * to support a pri as pri_net (as opposed to pri_cpe), either to talk to another * system or a PBX of some sort, I would be very interested in hearing about your experiences. Imparticular, I would like to know that it works before I invest in the extra hardware. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiki TOS - worrying for an open source project?
I've made a couple of small contributions to the wiki but recently I read the Terms of service, they are pretty draconian: Download (other than page caching), or modify this site. Reproduce, duplicate, copy, sell, resell, visit or use for other commercial purposes this site or any portion thereof. Use frames or framing techniques to enclose this site or any portion thereof for commercial purposes. Use meta tags or other 'hidden text' utilizing voip-info.org's name or trademarks. Any unauthorized use terminates the permission or license granted by voip-info.org. Sounds pretty damned decent to me so far... When you enter content into any area of this web site, unless stated otherwise, you grant voip-info.org and its affiliates a nonexclusive, royalty-free, perpetual, irrevocable, and fully sublicensable right to use, reproduce, modify, adapt, publish, translate, create derivative works from, distribute, and display such content throughout the world in any media. This is generally implied with any forum or newsgroup -- It's just restating what the above list said: you can't post publically and then turn around and sue voip-info.org for doing something with that information. You will note it said NONexclusive ... right -- they're not saying the information belongs to them, they are saying that the information is in the public domain. What worries me most is that the current terms seem crafted so as to ensure that should the people who run voip-info ever decide to remove content, or stop hosting the wiki, it couldn't be mirrored anywhere else. Untrue. Their terms about relinking or republishing are for COMMERCIAL use, unless I'm misreading something here. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Immortal SIP NAT problem
Ignace CARIA wrote: I know I know this subject have been The most written subject about VoIP :-) If Asterisk is on a Public IP Address and a softphone behind the nat, sip.conf must contains for this phone: nat=yes And in most cases qualify=yes The nat=yes makes asterisk don't trust the phone's information in regards to the IP address it comes from and the IP address it want's RTP sound to be sent to. Asterisk instead directs the signalling and media to the address we receive the packets from. For RTP, we also send the sound to the port we receive from (symmetric RTP). Note: If you're using an outbound proxy (IX66, SER) this will not work. Then it's the proxy's problem to sort out. IX66 is an excellent choice for this. So if this is your network configuration, don't turn on nat=yes. You will still need Symmetric RTP in most cases, so in chan_sip2 I've added a setting called symmetricrtp=yes that doesn't change the behaviour in regards to where we send SIP signalling, but change the behaviour of the media stream. I haven't gotten much feedback on this addition, but have good use of it myself. The qualify=yes sends out small packets to the client to measure the round-trip time for sending UDP packets. This is actually quite nice data to have, so you see how fast or slow link you have between the phone and your Asterisk server. An effect of sending those packets is that the NAT box keeps the session open, since we're actually communicating. That way, the session will be open when we signal that there's an incoming phone call. If the NAT doesn't get any keep-alive packets like this, it will close for business and there's no way we can open a call into the phone on the inside. Now if I want to configure my sipphone (X-Lite) placing behing the NAT, it must have in Domain/Realm the external IP address? No. Set Domain and realm to what it should be. Realm should match the realm= setting in your sip.conf, which should be globally unique. Your domain name or the hostname of the server is a good choice. With X-lite, in most cases you don't need to do anything special for NAT traversal. It has in itself an excellent support for NAT traversal, so you don't have to turn on nat=yes. It also sends it's own NAT keep-alives, so qualify= isn't needed. But even if Xlite is a wizard, your NAT device may be a disgusting beast. If Xlite doesn't work with your NAT, then change the status of Send internal IP. If Asterisk is behind the NAT, sip.conf must have in [globals] externip = External IP address localnet = Internal NETWORK address localmask = mask of localnet These settings is only needed for Asterisk when Asterisk is behind a NAT, registering with another SIP service provider on the outside. I would love seeing a good document, but the myriad of settings in various equipments and the behaviour av all different NAT's out there makes it very hard. Luckily, more and more vendors are starting to understand how STUN can help their equipment behave better. And new NAT boxes is better at handling this, so in most cases NAT=yes or a smart device, like Xten Xlite, with STUN support and some SIP header mangling magic, fixes this. Xten Xlite is really good at supporting STUN and DNS srv, so if you have configured your DNS right for your domain, clients will connect just by configuring domain, username and password. It will find the proxy and your stun server by looking up SRV records. It will figure out the workings of your NAT device and send the right signalling to the proxy. And no, I'm in no way affiliated with Xten Networks, inc. I'm just a happy user of the software. Ah, and of course, there is a good document with a lot of links on the topic of SIP and NAT. On the wiki, of course :-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Time to lock down v1.1?
Isn't it about time to lock down added functionality to v1.1 and fix the remaining bugs? There has been a significant amount of traffic on the cvs list, the irc and other channels with folks spending time adding new functionality to Head. Think its time to lock it down, fix the bugs that have been introduced, and get to something that the _majority_ can agree to call v1.1 Stable in real production terms. It's a known fact that bugs are not being fixed in Stable, and even Mark has suggested no one should be running Stable in a production environment. There has been a number of postings in the last few days relative to bugs in sip, iax2, zaptel, codecs, etc. The add-on folks are obviously also having problems keeping up with modifying patches to a constantly moving target, and applying those to Stable is fruitless. I'd even suggest that no v1.2 Head be created until such time as the majority of bugs are fixed, and that souce _then_ copied to whatever the next version is going to be called. All in favor? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] generate dial tone
Michael George [EMAIL PROTECTED] wrote: I did take a quick look at it, but the header indicated that DISA allows incoming calls to dial back out. I am just trying to emulate the feel of our current PBX which will just connect us to an outgoing line (with a dialtone) when we hit 9. (Though I don't want asterisk to mimic that behavior because I want to be judicious about which outgoing channels are used depending on the number dialed.) I've been reading your requirements as if you wanted an IVR system and wanted incoming users to be able to select '9' to get a new dial tone and dial out. It just occurred to me that perhaps what you really want is for your internal users to get a secondary dial tone (different than the primary) when they press '9'. Cisco phones allow this as an option in their dialplan. I forget now, but it may be a comma after the '9'. Other clients may allow this too (I don't think the Sipura SPA-2000 does, btw). I don't use that facility myself, as the secondary tone sounds terrible on Cisco phones. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Downgrading Asterisk
Although many of us that have worked in a production I/T arena assume something called Stable would truly have known bugs fixed, that's hardly the case for *. That branch really should be renamed to something like v1.0 and remove any reference to Stable and bug fixes as its treated as a lockdown for added functionality, and has nothing to do with functional stability. This comment shows you suffer from not understanding that words have more than one meaning. Stable means not changing much. A stable table doesn't fall over and not that it doesn't have flaws in the design such as being only 1 foot off of the ground. Similar people have the same mistaken opinion about Debian, it is stable because it doesn't change much. Only things that must change(security) gets changed in stable. Someone who runs stable shouldn't have to worry too much about things changing. Remember the reason for stable, it is there to make a run at a 1.0 code release. What software do you know of besides Hello World has a bug free 1.0 release. Critch, That's not even close to reasonable comments, and even Mark has made comments that contradict yours above. Stable might not roll over, but that's about all that can be said about it. We'll take it up off list if you really want to discuss it. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Time to lock down v1.1?
Hi Rich, Sounds like a good idea. Umar --- Rich Adamson [EMAIL PROTECTED] wrote: Isn't it about time to lock down added functionality to v1.1 and fix the remaining bugs? There has been a significant amount of traffic on the cvs list, the irc and other channels with folks spending time adding new functionality to Head. Think its time to lock it down, fix the bugs that have been introduced, and get to something that the _majority_ can agree to call v1.1 Stable in real production terms. It's a known fact that bugs are not being fixed in Stable, and even Mark has suggested no one should be running Stable in a production environment. There has been a number of postings in the last few days relative to bugs in sip, iax2, zaptel, codecs, etc. The add-on folks are obviously also having problems keeping up with modifying patches to a constantly moving target, and applying those to Stable is fruitless. I'd even suggest that no v1.2 Head be created until such time as the majority of bugs are fixed, and that souce _then_ copied to whatever the next version is going to be called. All in favor? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger - Communicate instantly...Ping your friends today! Download Messenger Now http://uk.messenger.yahoo.com/download/index.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Changes???
Hello Olle! Please add a SIP debug of the call so we can see what happens, who refuses what call. Situation: I'm behind an NAT firewall and get an incoming call from my SIP provider. I have the following entries in sip.conf: register = 1838933:[EMAIL PROTECTED]/1838933 [sipgate.de] type=user context=in-sip nat=1 language=de disallow=all allow=gsm allow=alaw allow=ulaw Unforunately not using an URL as name for the section as recommended does not work. Registration with my provider will fail because no section can be found so I used this one where this failure does not appear: == Parsing '/etc/asterisk/sip.conf': == Parsing '/etc/asterisk/sip.conf': Found May 28 16:47:01 WARNING[1114610608]: chan_sip.c:2191 sip_register: Host 'sipgate-in' not found at line 28 Here you are with my complete debugging information for an incoming call: - INVITE sip:MyNumber@172.20.0.2 SIP/2.0 Max-Forwards: 20 Record-Route: sip:MyNumber@217.10.79.9;ftag=40b74c2525f79;lr=on Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKc5ab.dcdaac71.0 Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKc5ab.ccdaac71.0 To: sip:MyNumber@sipgate.de From: sip:CallerID@sipgate.de;tag=40b74c2525f79 CSeq: 1 INVITE Call-ID: 40b74c2525f79.fifouacctd Content-Length: 155 User-Agent: Sip EXpress router(0.8.12-tcp_nonb (i386/linux)) Contact: sip:[EMAIL PROTECTED]:5060 Content-Type: application/sdp Sipgate-Authentication: accepted v=0 o=click-to-dial 0 0 IN IP4 0.0.0.0 s=session c=IN IP4 0.0.0.0 b=CT:1000 t=0 0 m=audio 40814 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=direction:active 14 headers, 9 lines Using latest request as basis request Sending to 217.10.79.9 : 5060 (non-NAT) Found RTP audio format 0 Peer RTP is at port 0.0.0.0:0 Found description format PCMU Capabilities: us - 0x40e(GSM|ULAW|ALAW|ILBC), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) Found peer 'sipgate-out' Reliably Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKc5ab.dcdaac71.0;received=217.10.79.9 Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKc5ab.ccdaac71.0 From: sip:CallerID@sipgate.de;tag=40b74c2525f79 To: sip:MyNumber@sipgate.de;tag=as0ce31626 Call-ID: 40b74c2525f79.fifouacctd CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:MyNumber@172.20.0.2 Proxy-Authenticate: Digest realm=voyager.localserver.de, nonce=2cb0b193 Content-Length: 0 to 217.10.79.9:5060 Scheduling destruction of call '40b74c2525f79.fifouacctd' in 15000 ms zion*CLI Sip read: ACK sip:MyNumber@172.20.0.2 SIP/2.0 Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKc5ab.dcdaac71.0 From: sip:CallerID@sipgate.de;tag=40b74c2525f79 Call-ID: 40b74c2525f79.fifouacctd To: sip:MyNumber@sipgate.de;tag=as0ce31626 CSeq: 1 ACK User-Agent: Sip EXpress router(0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 8 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 172.20.0.2:5060;branch=z9hG4bK1eb6d6a5 From: sip:MyNumber@sipgate.de;tag=as166899a7 To: sip:MyNumber@sipgate.de Call-ID: [EMAIL PROTECTED] CSeq: 109 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:MyNumber@172.20.0.2 Event: registration Content-Length: 0 (no NAT) to 217.10.79.9:5060 zion*CLI Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.20.0.2:5060;branch=z9hG4bK1eb6d6a5 From: sip:MyNumber@sipgate.de;tag=as166899a7 To: sip:MyNumber@sipgate.de;tag=b11cb9bb270104b49a99a995b8c68544.94c0 Call-ID: [EMAIL PROTECTED] CSeq: 109 REGISTER WWW-Authenticate: Digest realm=sipgate.de, nonce=40b74d5fe384afdade9e26b4da34a52421ad4140 Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 Warning: 392 217.10.79.9:5060 Noisy feedback tells: pid=15717 req_src_ip=172.20.0.2 req_src_port=5060 in_uri=sip:sipgate.de out_uri=sip:sipgate.de via_cnt==1 10 headers, 0 lines 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 172.20.0.2:5060;branch=z9hG4bK2a07a252 From: sip:MyNumber@sipgate.de;tag=as166899a7 To: sip:MyNumber@sipgate.de Call-ID: [EMAIL PROTECTED] CSeq: 110 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=MyNumber, realm=sipgate.de, algorithm=MD5, uri=sip:sipgate.de, nonce=40b74d5fe384afdade9e26b4da34a52421ad4140, response=17d50f31e37949b4dd8e65e91f6c5002, opaque= Expires: 120 Contact: sip:MyNumber@172.20.0.2 Event: registration Content-Length: 0 (no NAT) to 217.10.79.9:5060 zion*CLI Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.20.0.2:5060;branch=z9hG4bK2a07a252 From: sip:MyNumber@sipgate.de;tag=as166899a7 To: sip:MyNumber@sipgate.de;tag=b11cb9bb270104b49a99a995b8c68544.2876 Call-ID: [EMAIL PROTECTED] CSeq: 110 REGISTER Contact: sip:MyNumber@172.20.0.2;q=0.00;expires=120 Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 Warning: 392 217.10.79.9:5060 Noisy feedback tells:
[Asterisk-Users] Re: [Asterisk-Dev] Time to lock down v1.1?
Rich Adamson wrote: It's a known fact that bugs are not being fixed in Stable, and even Mark has suggested no one should be running Stable in a production environment. On the other hand, there's not many bugs open in the bug tracker. Feature requests and patches, but not bugs. If you are aware of bugs in stable or head, please report them a.s.a.p. so we can start fixing them. Life as a bug marshal has been quite easy for a while, with Mark fixing bugs like crazy and not many new bugs being reported. I guess you do not want the bug marshals to fall asleep and live a bug-free life :-) 1.0 will be the stable release. There hasn't been many fixes to that one lately, only MAJOR bug fixes has been applied. It will not be relased according to any plan, remember - this is Open Source. It will be released when considered stable with no open bugs. 1.1 (today's head) is more of a let's try if this works' release. Please spend time testing it. Remember, CVS HEAD, is not meant to be stable. Now and then, it might not even compile cleanly. It's a developer's release, at some point in future aimed to be stable. And, as always, when reporting, don't forget to report which version you are running, on which platform. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Time to lock down v1.1?
Rich Adamson [EMAIL PROTECTED] wrote: Isn't it about time to lock down added functionality to v1.1 and fix the remaining bugs? There has been a significant amount of traffic on the cvs list, the irc and other channels with folks spending time adding new functionality to Head. Think its time to lock it down, fix the bugs that have been introduced, and get to something that the _majority_ can agree to call v1.1 Stable in real production terms. It's a known fact that bugs are not being fixed in Stable, and even Mark has suggested no one should be running Stable in a production environment. There has been a number of postings in the last few days relative to bugs in sip, iax2, zaptel, codecs, etc. The add-on folks are obviously also having problems keeping up with modifying patches to a constantly moving target, and applying those to Stable is fruitless. I'd even suggest that no v1.2 Head be created until such time as the majority of bugs are fixed, and that souce _then_ copied to whatever the next version is going to be called. All in favor? I'm in favour of that. Make it so. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 channel bank problem
Hi all. I have and E1 channel bank from Loop Telecom. there's a little issue with it, I cannot ring the phones on fxs interface, but can connect without issue them. What happens: I dial the phone on port 1, asterisk says Zap/1 is ringing, but the phone on the analog port doesn't ring. but if I take off hook the ringed phone, asterisk detects the answer at they're bridged correctly. also I can flash transfer without probs. only ring doesn't work. doing the ring test from the channel bank test menu, is all ok: the phones ring without issues. zaptel.conf says: span = 1,1,0,cas,hdb3,crc4 fxoks = 1-31 loadzone = us defaultzone = us zapata.conf is simply transfer=yes echocancel=yes threewaycalling=yes signalling=fxo_ks context=interni channel=1-31 any hint on where I can search for problems? -- Matteo Brancaleoni [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiki TOS - worrying for an open source project?
On Fri, 2004-05-28 at 08:37, Andrew Kohlsmith wrote: Please do not trim out attribution tags. The double quoted is from Julien Levi [EMAIL PROTECTED] What worries me most is that the current terms seem crafted so as to ensure that should the people who run voip-info ever decide to remove content, or stop hosting the wiki, it couldn't be mirrored anywhere else. Untrue. Their terms about relinking or republishing are for COMMERCIAL use, unless I'm misreading something here. The other part is that a wiki is really unmirrorable using normal methods of mirroring a site. You need to just run the same software and have the database behind it mirrored. I'm sure if the wiki is running a new enough version of mysql, and the admin is willing, you could set up a mirror of the database and then set up a full on replication. Mysql supposedly supports replication, might want to put it to some use. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Database
I'd like to be able to add additional fields to the the Asterisk database. I'm using Mysql for most of my data lookup and manipulation, and it seems to work pretty well. In keeping with what I know how to do, it would be very handy to be able to insert say a call forward number into a customer record. That way, I could automatically route calls to extensions to a forwarded number. Any suggestions on how this can be done? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiki TOS - worrying for an open source project?
Please do not trim out attribution tags. The double quoted is from Julien Levi [EMAIL PROTECTED] Why not? I replied to Julien Levi's post, so the attribution should be implied, just as I am replying to your post, and I don't have a Steven Critchfield sez line... I've been doing this for damn near a decade now and you're the first person I can recall making mention of it. If I'm quoting mulitple levels or multiple people, I will of course try to make the attribution clear, but for this simple stuff, I thought it was already. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: * as pri_net?
We are using Asterisk as pri_net connected to Merlin Legeng over DS100 card. It works quite stable and did not see any problem for past months. Here is my configs: === /etc/zaptel.conf # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # span=1,0,2,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us === /etc/asterisk/zapata.conf [channels] language=en context=default switchtype=national pridialplan=private overlapdial=no signalling=pri_net usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=yes callreturn=yes echocancel=32 echocancelwhenbridged=yes echotraining=yes rxgain=2.5 txgain=2.5 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no ; progzone=us musiconhold=default channel = 1-23 Mark Johnston wrote: Bruce Komito [EMAIL PROTECTED] wrote: If you have used * to support a pri as pri_net (as opposed to pri_cpe), either to talk to another * system or a PBX of some sort, I would be very interested in hearing about your experiences. I've used pri_net on one end to talk between two Asterisk boxes with T100Ps and a T1 crossover cable. It worked exactly as advertised - just make sure the span= statements in zaptel.conf are right (one span=1,1,0,esf,b8zs and the other span=1,0,0,esf,b8zs). Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards, Vasyl Rublyov IonIdea, Inc. 3913, Old Lee Highway, Suite 33B Fairfax, VA 22030 Tel: (703) 691-0400 Mob: (703) 395-0238 Fax: (703) 691-0401 www.ionidea.com A CMM Level III and ISO 9001 Company - This e-mail (including any attachments) is confidential and may be legally privileged. If you are not an intended recipient or an authorized representative of an intended recipient, you are prohibited from using, copying or distributing the information in this e-mail or its attachments. If you have received this e-mail in error, please notify the sender immediately byreturn e-mail and delete all copies of this message and any attachments.
[Asterisk-Users] SIP Registration Problem
Title: Re: [Asterisk-Users] Wiki TOS - worrying for an open sourceproject? I am using snom200 phones registering with Asterisk via SIP. I can see where the phone registers without a problem, and then when you try and make a call I get a proxy authentication required message on the phone and failed to authenticate user error in the Asterisk messages file. Then the next call you make from the phone goes through without a problem. Nothing changes between these two events, but it is almost like the phone is using two different passwords for the same account. Has anyone else seen a problem like this? I am using an Asterisk CVS version from early March, not sure if upgrading will help as well. Thanks, Brian
RE: [Asterisk-Users] * as pri_net?
I've done this too. Four E1's on one box, talking to four E1's on another asterisk box. I just use it for load testing new Zap versions. Note that you need a crossover E1 cable for this. Cheers Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Komito Sent: Friday, May 28, 2004 6:35 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] * as pri_net? If you have used * to support a pri as pri_net (as opposed to pri_cpe), either to talk to another * system or a PBX of some sort, I would be very interested in hearing about your experiences. Imparticular, I would like to know that it works before I invest in the extra hardware. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with PPP internet T1
Hello all, We have a TE405P set up with span 1 running to a channel bank, a PRI running into span 2, and a PPP internet T1 running into span 3. We have the first 2 spans up and running without a problem. We have hdlc compiled into the kernel and after making the appropriate changes to zaptel.conf and loading the zaptel, wct4xxp, and hldc modules we can bring up the third span with the internet T1, but we can't seem to communicate with the ISP. We ran the following commands: sethdlc hdlc0 ppp ifconfig hdlc0 our serial ip pointopoint isp gateway ip netmask isp subnet mask -arp Now we can ping our serial ip, but can't ping the isp gateway ip. ifconfig shows us transmitting packets, but we don't receive any. Any help would be greatly appreciated. Thanks, Patrick -- This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice Pulse
On Thu, 2004-05-27 at 22:07, Aaron J. Angel wrote: Did you know that by clicking reply, one is following proper netiquette? It is especially helpful for those using threaded mail readers. On top of that, if people delete messages simply because they don't like the subject, who's problem is that? Unless, of course, the person is writing about NOTHING that has to do with the original thread. That was the case here. The Subject: was Voicepulse, but his questions were about specific non-Voicepulse Astersik issues -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Changes???
The failure has just been fixed as I saw in mantis: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001738 Thanks a lot! ;D Regards Julian Pawlowski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiki TOS - worrying for an open source project?
Hi! I've made a couple of small contributions to the wiki but recently I read the Terms of service, they are pretty draconian: [...] What worries me most is that the current terms seem crafted so as to ensure that should the people who run voip-info ever decide to remove content, or stop hosting the wiki, it couldn't be mirrored anywhere else. I share your concerns, should have looked at the terms earlier - so Arte Marketing can at any moment run away with all my contributions and close the site, and the only currently workable documentation for Asterisk is lost ... !? Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk addons
On Fri, 28 May 2004, CW_ASN wrote: - Original Message - From: Fabio Donaggio To: [EMAIL PROTECTED] Sent: Friday, May 28, 2004 6:16 AM Subject: [Asterisk-Users] Asterisk addons Hi to all!! Is there another method to download asterisk addons??? Thanks F Man! Try to investigate for yourself! Use google! http://www.google.com/search?q=asterisk-addons+downloadie=UTF-8hl=esmeta = As a side note, someone approached me and mentioned the possibility of sponsoring me monetarily to build Asterisk-AddOns and Asterisk-Sounds RPMS for the community. Are there other people interested in this? It would probably be done relatively quickly, but I'd need some additional contributions to justify to my partner taking time away from putting food on the table to focus on it and get it done. Email me off list if you are interested. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * as pri_net?
On Fri, 2004-05-28 at 08:34, Bruce Komito wrote: If you have used * to support a pri as pri_net (as opposed to pri_cpe), either to talk to another * system or a PBX of some sort, I would be very interested in hearing about your experiences. Imparticular, I would like to know that it works before I invest in the extra hardware. I'm glad to see all the success stories here, as I am so far the only failure so far and it is explainable. According to JerJer, I will have to upgrade the libpri on one of my machines to get it to work properly. Once that is done, it will be fine. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: Asterisk and MySQL
Hi! It's all ok with CVS login...I download asterisk-addons. I would try to store sip friends in MySQL database and also the voicemailcan you help me??? Thanks
Re: [Asterisk-Users] Conference Server
HI there, Thanks everybody for all the answers. I took a look at the asterisk timer ztdummy page (http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy) Unfortunaly, my PC has the USB OHCI module. So, I downloaded the zaprtc module from http://www.junghanns.net/asterisk/. I tried to do make, and got the following error message: [EMAIL PROTECTED] zaptelrtc]# make cc -c zaprtc.c -D__KERNEL__ -DMODULE -DEXPORT_SYMTAB -fomit-frame-pointer -O2 -Wall -I/usr/src/linux/include -Wall En el fichero includo de /usr/include/linux/module.h:20, de zaprtc.c:60: /usr/include/linux/modversions.h:1:2: #error Modules should never use kernel-headers system headers, /usr/include/linux/modversions.h:2:2: #error but rather headers from an appropriate kernel-source package. /usr/include/linux/modversions.h:3:2: #error Change -I/usr/src/linux/include (or similar) to /usr/include/linux/modversions.h:4:2: #error -I/lib/modules/$(uname -r)/build/include /usr/include/linux/modversions.h:5:2: #error to build against the currently-running kernel. In file included from /usr/include/linux/sched.h:14, from /usr/include/linux/mm.h:4, from /usr/include/linux/locks.h:5, from /usr/include/linux/devfs_fs_kernel.h:6, from /usr/include/linux/miscdevice.h:4, from zaprtc.c:63: /usr/include/linux/timex.h:56: error: error sintctico before and In file included from /usr/include/linux/timex.h:126, from /usr/include/linux/sched.h:14, from /usr/include/linux/mm.h:4, from /usr/include/linux/locks.h:5, from /usr/include/linux/devfs_fs_kernel.h:6, from /usr/include/linux/miscdevice.h:4, from zaprtc.c:63: /usr/include/asm/timex.h:33: error: error sintctico before cacheflush_time /usr/include/asm/timex.h:35: error: error sintctico before get_cycles In file included from /usr/include/linux/sched.h:14, from /usr/include/linux/mm.h:4, from /usr/include/linux/locks.h:5, from /usr/include/linux/devfs_fs_kernel.h:6, from /usr/include/linux/miscdevice.h:4, from zaprtc.c:63: /usr/include/linux/timex.h:147: error: field `time' has incomplete type En el fichero includo de /usr/include/linux/bitops.h:69, de /usr/include/asm/system.h:7, de /usr/include/linux/sched.h:16, de /usr/include/linux/mm.h:4, de /usr/include/linux/locks.h:5, de /usr/include/linux/devfs_fs_kernel.h:6, de /usr/include/linux/miscdevice.h:4, de zaprtc.c:63: /usr/include/asm/bitops.h:327:2: aviso: #warning This includefile is not available on all architectures. /usr/include/asm/bitops.h:328:2: aviso: #warning Using kernel headers in userspace: atomicity not guaranteed In file included from /usr/include/linux/signal.h:4, from /usr/include/linux/sched.h:25, from /usr/include/linux/mm.h:4, from /usr/include/linux/locks.h:5, from /usr/include/linux/devfs_fs_kernel.h:6, from /usr/include/linux/miscdevice.h:4, from zaprtc.c:63: /usr/include/asm/signal.h:107: error: error sintctico before sigset_t /usr/include/asm/signal.h:110: error: error sintctico before '}' token In file included from /usr/include/linux/sched.h:81, from /usr/include/linux/mm.h:4, from /usr/include/linux/locks.h:5, from /usr/include/linux/devfs_fs_kernel.h:6, from /usr/include/linux/miscdevice.h:4, from zaprtc.c:63: /usr/include/linux/timer.h:45: error: error sintctico before spinlock_t /usr/include/linux/timer.h:53: error: error sintctico before '}' token /usr/include/linux/timer.h:67: error: error sintctico before tvec_base_t /usr/include/linux/timer.h:101: error: error sintctico before tvec_bases /usr/include/linux/timer.h: En la funcin `init_timer': /usr/include/linux/timer.h:105: error: dereferencing pointer to incomplete type /usr/include/linux/timer.h:105: error: dereferencing pointer to incomplete type /usr/include/linux/timer.h:106: error: dereferencing pointer to incomplete type /usr/include/linux/timer.h: En la funcin `timer_pending': /usr/include/linux/timer.h:121: error: dereferencing pointer to incomplete type En el fichero includo de /usr/include/linux/devfs_fs_kernel.h:6, de /usr/include/linux/miscdevice.h:4, de zaprtc.c:63: /usr/include/linux/locks.h:8:27: linux/pagemap.h: No existe el fichero o el directorio In file included from /usr/include/linux/devfs_fs_kernel.h:6, from /usr/include/linux/miscdevice.h:4, from zaprtc.c:63: /usr/include/linux/locks.h: En la funcin `wait_on_buffer': /usr/include/linux/locks.h:19: error:
Re: [Asterisk-Users] Re: [Asterisk-Dev] Time to lock down v1.1?
On Fri, 28 May 2004, Olle E. Johansson wrote: Rich Adamson wrote: It's a known fact that bugs are not being fixed in Stable, and even Mark has suggested no one should be running Stable in a production environment. On the other hand, there's not many bugs open in the bug tracker. Feature requests and patches, but not bugs. If you are aware of bugs in stable or head, please report them a.s.a.p. so we can start fixing them. Life as a bug marshal has been quite easy for a while, with Mark fixing bugs like crazy and not many new bugs being reported. I guess you do not want the bug marshals to fall asleep and live a bug-free life :-) 1.0 will be the stable release. There hasn't been many fixes to that one lately, only MAJOR bug fixes has been applied. It will not be relased according to any plan, remember - this is Open Source. It will be released when considered stable with no open bugs. Wew.. after reading the last post, I had to stop and think if I was going crazy or not! Just for the record and to make sure that my understanding is correct, 1.0 is frozen and no NEW features are being added to that tree, correct? Aside from Major fixes, 1.0 is very near a release candidate. I might suggest that some of the IAX2 and SIP bugs (RTP Timestamps, etc..) be applied to Stable (for all I know they already might be) but otherwise, we start moving towards a 1.0-rc1 archive. I'll RPM up whatever you guys decided to drop, and continue to run 1.0_stable on my production boxes and provide feedback to the Bug Marshalls. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiki TOS - worrying for an open source project?
Andrew Kohlsmith wrote: Please do not trim out attribution tags. The double quoted is from Julien Levi [EMAIL PROTECTED] Why not? I replied to Julien Levi's post, so the attribution should be implied, just as I am replying to your post, and I don't have a Steven Critchfield sez line... I've been doing this for damn near a decade now and you're the first person I can recall making mention of it. Actually, it may be implied by you but it is not for the rest of the world. If I'm quoting mulitple levels or multiple people, I will of course try to make the attribution clear, but for this simple stuff, I thought it was already. Your mailreader should make the attribution for you. You don't need to try to make it clear. Just stop snipping the attribution manually. John Been doing this for over two decades now Fraizer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with PPP internet T1
What is your kernel version? Patrick J. Conroy wrote: Hello all, We have a TE405P set up with span 1 running to a channel bank, a PRI running into span 2, and a PPP internet T1 running into span 3. We have the first 2 spans up and running without a problem. We have hdlc compiled into the kernel and after making the appropriate changes to zaptel.conf and loading the zaptel, wct4xxp, and hldc modules we can bring up the third span with the internet T1, but we can't seem to communicate with the ISP. We ran the following commands: sethdlc hdlc0 ppp ifconfig hdlc0 our serial ip pointopoint isp gateway ip netmask isp subnet mask -arp Now we can ping our serial ip, but can't ping the isp gateway ip. ifconfig shows us transmitting packets, but we don't receive any. Any help would be greatly appreciated. Thanks, Patrick -- Thanks and regards, Vasyl Rublyov IonIdea, Inc. 3913, Old Lee Highway, Suite 33B Fairfax, VA 22030 Tel: (703) 691-0400 Mob: (703) 395-0238 Fax: (703) 691-0401 www.ionidea.com A CMM Level III and ISO 9001 Company - This e-mail (including any attachments) is confidential and may be legally privileged. If you are not an intended recipient or an authorized representative of an intended recipient, you are prohibited from using, copying or distributing the information in this e-mail or its attachments. If you have received this e-mail in error, please notify the sender immediately byreturn e-mail and delete all copies of this message and any attachments. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Development SOP - was:Downgrading Asterisk
I'm willing to open my system up for those developers that cannot duplicate the problem on their own systems. I have a nice flat network, good hardware, no off-the-wall configurations, an up-to-date kernel and server hardware, etc. Contact me on or off list and I'll arrange for SSH access for you, after we have a short phone conversation. Nik Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, May 28, 2004 7:59 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Downgrading Asterisk The code changes that fixed the cisco choppy sound for Stable went in last Friday. That change corrected iax2 issues that had been known for well over a month but never got applied to Stable. That same code is in Head, however many other changes have happened to Head, and some of those apparently have impacted at least some of us (mostly cisco users). Stable has a number of other bugs that reportedly will never get fixed as the fixes use functionality that exists only in Head. It seems the choppy (and almost unusable) audio in Head is only impacting some cisco users, and since these problems are not impacting the few that can read code, use cisco phones, and are impacted, we're stuck with the problem. The problem seems to be very evasive, however switching the iax2 links to use only iLBC (and not gsm) has corrected issues for some. Although many of us that have worked in a production I/T arena assume something called Stable would truly have known bugs fixed, that's hardly the case for *. That branch really should be renamed to something like v1.0 and remove any reference to Stable and bug fixes as its treated as a lockdown for added functionality, and has nothing to do with functional stability. FYI Downgrading to -stable totally fixed the choppy audio on Cisco my 7960 - * - IAX setup. Now, when would a fix that goes into stable get into the current source (HEAD)? And, isn't checking stuff into a stable branch that doesn't exist elsewhere in the source tree break some rules somewhere? It has to. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nik Martin Sent: Tuesday, May 25, 2004 2:53 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Downgrading Asterisk I upgraded to the latest HEAD version of asterisk, and all IAX calls started sounding choppy. It was suggested on the IRC channel that I go back to asterisk -stable to determine if that fixes it. Is downgrading as simple as upgrading? Because now, -stable builds fine, but I get an error on the asterisk console when starting, something about ast_get_txt not found. Recompiling and installing asterisk HEAD afterwards works just fine. As a side note, I recently upgraded my kernel to 2.4.26 and had an issue with old kernel headers, but have since resolved that prior to trying this downgrade. Any ideas? Nik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiki TOS - worrying for an open source project?
From: Steven Critchfield [EMAIL PROTECTED] The other part is that a wiki is really unmirrorable using normal methods of mirroring a site. You need to just run the same software and have the database behind it mirrored. I'm sure if the wiki is running a new enough version of mysql, and the admin is willing, you could set up a mirror of the database and then set up a full on replication. Mysql supposedly supports replication, might want to put it to some use. -- I don't know who is hosting the Wiki right now, but we are willing to either host the Wiki as a mirror, or be a mysql replication mirror. We are using mysql replication right now to replicate amongst three servers for our RADIUS and other hosted apps and it works very well. We also do daily backups of the master mysql server to an offsite location. We would do this free of charge, of course. We are using asterisk as a media gateway with Digiums TE405P cards and we appreciate the work that is going into Asterisk. Contact [EMAIL PROTECTED] or 972-617-2877 yours, Matthew Simpson TxLink Communications ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Downgrading Asterisk
The disconnect between HEAD and stable is what concerns me. The fact that a fix was put into Stable for the choppy audio on Cisco -*-IAX that I couldn't find in HEAD, and that didn't work when fetching and rebuilding HEAD is what concerns me. If it exists in stable (and works in stable), but doesn't work in HEAD, I'm puzzled. I worked at a very large development shop whose software is used in mission critical public safety environments. Changes would NEVER go into a release marked STABLE (and that were consequently feature locked and bug fix locked) that weren't extensively tested in the current development release. These changes would go into the NEXT STABLE release, unless they were a show stopper type bug. Also, diffing the current HEAD between just a few revisions makes me quite nervous, as a product that already has a STABLE branch shouldn't be showing as much feature creep as this one does. But, it's Open Source, and that's what you get sometimes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, May 28, 2004 8:14 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Downgrading Asterisk On Fri, 2004-05-28 at 07:59, Rich Adamson wrote: Although many of us that have worked in a production I/T arena assume something called Stable would truly have known bugs fixed, that's hardly the case for *. That branch really should be renamed to something like v1.0 and remove any reference to Stable and bug fixes as its treated as a lockdown for added functionality, and has nothing to do with functional stability. This comment shows you suffer from not understanding that words have more than one meaning. Stable means not changing much. A stable table doesn't fall over and not that it doesn't have flaws in the design such as being only 1 foot off of the ground. Similar people have the same mistaken opinion about Debian, it is stable because it doesn't change much. Only things that must change(security) gets changed in stable. Someone who runs stable shouldn't have to worry too much about things changing. Remember the reason for stable, it is there to make a run at a 1.0 code release. What software do you know of besides Hello World has a bug free 1.0 release. Please watch the inflammatory tone of your message next time you criticize the free software you are using and the people giving you their time. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Receptionist manager program.
We are writing a program using the manager for * for our receptionist to use once the system go live. If anyone is interested in helping us with testing please let me know. We are designing it for a touch screen monitor for her to do transfers, see whose on the phone and a few other features. Its in the development stage and has bugs. but I think its gonna be really good. If your interested please let me know. Im gonna be putting up a site for downloading if there is enough interest. We are considering writing a SIP client build into the program at a later time. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * as pri_net?
Bruce Komito wrote: If you have used * to support a pri as pri_net (as opposed to pri_cpe), either to talk to another * system or a PBX of some sort, I would be very interested in hearing about your experiences. Imparticular, I would like to know that it works before I invest in the extra hardware. TIA Bruce Komito Here too asterisk/TE410P ISDN-PRI TN767E/Definity G3si v6 switchtype = 5ess signalling = pri_net inbound/outbound, ext/ext, DNIS/ANI all working well. Very cool! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk addons
As a sidenote, your site doesn't work in Mozilla Firefox. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: [Asterisk-Dev] Time to lock down v1.1?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Friday, May 28, 2004 9:38 AM To: [EMAIL PROTECTED] Cc: Asterisk-a-users-list Subject: [Asterisk-Users] Re: [Asterisk-Dev] Time to lock down v1.1? On the other hand, there's not many bugs open in the bug tracker. Feature requests and patches, but not bugs. If you are aware of bugs in stable or head, please report them a.s.a.p. so we can start fixing them. In my observations, there are some personality conflicts on this list that make bug reporting difficult sometimes. People post to the list (prior to a formal bug-report in mantis) about a possible bug, and get chastised and berated because It works fine on my system, you must be doing something wrong, etc. After hearing that just so many times, You just give up and move on. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference Server
On Fri, 2004-05-28 at 10:53, pesb wrote: HI there, Thanks everybody for all the answers. I took a look at the asterisk timer ztdummy page (http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy) Unfortunaly, my PC has the USB OHCI module. So, I downloaded the zaprtc module from http://www.junghanns.net/asterisk/. I tried to do make, and got the following error message: [EMAIL PROTECTED] zaptelrtc]# make cc -c zaprtc.c -D__KERNEL__ -DMODULE -DEXPORT_SYMTAB -fomit-frame-pointer -O2 -Wall -I/usr/src/linux/include -Wall En el fichero incluído de /usr/include/linux/module.h:20, de zaprtc.c:60: /usr/include/linux/modversions.h:1:2: #error Modules should never use kernel-headers system headers, How can I install zaprtc on my PC. I have a PIV Fedora Core 1 with a 2.4.22-1.2115.nptl kernel? First, open your eyes and read the messages. Second use google. Google is there for just such a problem. Visit this url and marvel at how easy it is to ask google a question. http://tinyurl.com/2ajso I responded to a message not but half a month ago to tell the person to do the same thing. Install the kernel source. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] seeking an example for Message Waiting Indicator stutter dialtone
does anyone have an example they would please share for turning on stutter dialtone for a zaptel channel when there is a message waiting? Thanks! Paul Paul Mahler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * as pri_net?
I have digium E1s as pri_net connected to nms based softswitch - no problems Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bruce Komito Sent: Friday, May 28, 2004 3:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] * as pri_net? If you have used * to support a pri as pri_net (as opposed to pri_cpe), either to talk to another * system or a PBX of some sort, I would be very interested in hearing about your experiences. Imparticular, I would like to know that it works before I invest in the extra hardware. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Freenet iPhone w/Asterisk
The reason why I would like to use Freenet iPhone is their cheap rate for calls to Germany (1 cent/min). It is correct that you have to sign up for one of their DSL plans. But the pay as you go plan has neither monthly fee and nor a minimum usage requirement. The lack of incoming phone number / DID's is not a problem because I just want to do some least cost routing for calls to Germany. But back to my configuration issues. I figured out what the problems were ... First of all there were usual NAT/SIP issues - that I fixed (thanks for pointing me to the fwd web forum!). The other issue was a little trickier: Freenet uses a hostname iphone.freenet.de. So if I called the number '123456' - * would put [EMAIL PROTECTED] into the SIP messages. But Freenet expects something like [EMAIL PROTECTED] I could not find any way to configure that with *. So what I did was put an entry into my /etc/hosts configuration with host freenet.de and the actual IP address of iphone.freenet.de and change the * sip host entry to freenet.de instead of iphone.freenet.de. Now it works fine. But I wonder if there is any way to configure that in *? jo wrote: Oliver, you should be able to connect * with the same settings required for softphones. http://www.freenet.de/freenetiphone/sip_telefone/index.html Firewall problems depends on your individual situation, a search in this list or browsing fwd's web forum may find a solution. But why would you do that? freenet iPhone is a rather prorietary service without any gateways except PSTN (which is limited to freenet DSL users). They don't even offer DIDs. my1 cent jo [EMAIL PROTECTED] wrote: Has anybody tried to use Freenet's Germany based iPhone Service with Asterisk? Maybe even from behind a NAT? Freenet seems to use SER ... but I can not get a connection to their SIP proxy from Asterisk going through a NATed firewall. Asterisk -SIP- Firewall with NAT -SIP- Freenet iPhone server Thanks, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiki TOS - worrying for an open source project?
On Fri, 2004-05-28 at 10:23, Andrew Kohlsmith wrote: Please do not trim out attribution tags. The double quoted is from Julien Levi [EMAIL PROTECTED] Why not? I replied to Julien Levi's post, so the attribution should be implied, just as I am replying to your post, and I don't have a Steven Critchfield sez line... I've been doing this for damn near a decade now and you're the first person I can recall making mention of it. If I'm quoting mulitple levels or multiple people, I will of course try to make the attribution clear, but for this simple stuff, I thought it was already. You have to realize that not all users have threaded mail readers. Also that sometimes, but not on this list, list software will strip headers down to a point that threading can break. Think also of the person doing a search later on via the archives and they see your post without the others, having attribution will help in the chance case you trimmed something the person is looking for. Specifically they know better where to look backwards to find the cause for your message. Length of committing an action doesn't imply it is correct. Many people would have let it slide. Normally I probably would have skipped the point if I hadn't wanted to look backwards and verify an opinion. So I made the simple comment and even said please. You can search out my treatment of others who fall foul of nettiquette guidelines to see I was unusually polite here. As has already been mentioned, please don't trim the attribution out as it should be provided by your mail reader for you. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with PPP internet T1
We are using redhat 8 with kernel 2.4.18-14. We recompiled the kernel with the hdlc-2.4.20-1.14a.patch from http://hq.pm.waw.pl/hdlc/. That site stated that this was the patch to use for 2.4.20 and earlier kernels. The kernel seemed to compile and sethdlc seemed to compile fine and the hdlc module loads and we see the hdlc0 network device. Patrick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vasyl Rublyov Sent: Friday, May 28, 2004 12:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with PPP internet T1 What is your kernel version? Patrick J. Conroy wrote: Hello all, We have a TE405P set up with span 1 running to a channel bank, a PRI running into span 2, and a PPP internet T1 running into span 3. We have the first 2 spans up and running without a problem. We have hdlc compiled into the kernel and after making the appropriate changes to zaptel.conf and loading the zaptel, wct4xxp, and hldc modules we can bring up the third span with the internet T1, but we can't seem to communicate with the ISP. We ran the following commands: sethdlc hdlc0 ppp ifconfig hdlc0 our serial ip pointopoint isp gateway ip netmask isp subnet mask -arp Now we can ping our serial ip, but can't ping the isp gateway ip. ifconfig shows us transmitting packets, but we don't receive any. Any help would be greatly appreciated. Thanks, Patrick -- Thanks and regards, Vasyl Rublyov IonIdea, Inc. 3913, Old Lee Highway, Suite 33B Fairfax, VA 22030 Tel: (703) 691-0400 Mob: (703) 395-0238 Fax: (703) 691-0401 www.ionidea.com A CMM Level III and ISO 9001 Company - This e-mail (including any attachments) is confidential and may be legally privileged. If you are not an intended recipient or an authorized representative of an intended recipient, you are prohibited from using, copying or distributing the information in this e-mail or its attachments. If you have received this e-mail in error, please notify the sender immediately byreturn e-mail and delete all copies of this message and any attachments. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content, and is believed to be clean. -- This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] Time to lock down v1.1?
I'll RPM up whatever you guys decided to drop, and continue to run 1.0_stable on my production boxes and provide feedback to the Bug Marshalls. I'll do slackware 9.1 packages for anyone interested if there aren't any other maintainers... -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Changes???
Hi! The failure has just been fixed as I saw in mantis: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001738 Unfortunately that didn't solve my problem - however I am not sure anymore that this is related, and maybe I just have a basic misunderstanding concerning type=peer and type=user. Question: Why do I need type=peer for both cases, e.g. incoming AND outgoing calls? I am really confused here - or someone/something else is... ;- 1. I want to be able to dial out to FWD with a Dial() statement in extensions.conf that does not include username or password so that these do not show up in the CDRs, e.g. using Dial(SIP/[EMAIL PROTECTED]) 2. The above only works if FreeWorld-out-user1 is of type=peer (and not type=user) 3. On an incoming FWD call * unfortunately always matches the host to the [FreeWorld-out-user1] section instead of the [FreeWorld-incoming] section, which is kind of logic becase both are peers. Then authentication fails because the calling user naturally doesn't have the correct password for FreeWorld-out-user1. Cheers, Philipp [FreeWorld-incoming] context=from-FreeWorld type=peer host=fwd.pulver.com [FreeWorld-out-user1] type=peer secret= username=yy fromuser=yy host=fwd.pulver.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Asterisk with Draytek 2600V
Hello louis, Friday, May 28, 2004, 6:32:33 PM, you wrote: lg Hi Alessio lg Thank you for the reply. Our configuration is as follows lg Asterisk Server 192.168.0.250 is on our LAN lg Vigor 192.168.1.1 connects to the LAN VPN (vigor to vigor) lg Laptop 192.168.1.10 with XLite I can suggest this: turn off xlite on the laptop, then reset the vigor that's on the side of the laptop. I can guess it will work then, I found similar problems some time ago. It seems that the vigor voip ports are only working if there are no sip clients behind the ethernet port, maybe it's some kind of port redirection issue. I can also say that vigor support is, in my experience, quick and very helpful. Hope it helps ! lg The Vigor and Laptop both register with Asterisk using their correct private lg ip's ie 192.168.1.1 and 192.168.1.10 lg I can make and recieve calls fine on the Laptop but not on the Vigor. lg I have yet to try placing the Asterisk server on a public IP address but I lg may try this tomorrow when I am back in the office. Any ideas? I have a lg standard SIP.CONF with no special config options but I may be missing lg something. lg Many Thanks lg Louis Guadagno lg Network Manager lg Practical Law Company -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
On Fri, 28 May 2004, Tony Hoyle wrote: Actually it's the first time I've ever heard of distinctive ring being available in the UK... :) BT launched Call Sign sometime in 1996. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp wont compile.
I can't get spandsp to compile. when I go to the */apps directory i continually fails. Makefile:80: warning: overriding commands for target `app_rxfax.so' Makefile:77: warning: ignoring old commands for target `app_rxfax.so' cc -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [app_rxfax.o] Error 1 I chamged the Makefile to include app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_rxfax.so : app_rxfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_rxfax. o app_rxfax.c app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_txfax.o: app_txfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_txfax.o app_txfax.c any ideas? thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Time to lock down v1.1?
1.1 (today's head) is more of a let's try if this works' release. Please spend time testing it. Remember, CVS HEAD, is not meant to be stable. Now and then, it might not even compile cleanly. It's a developer's release, at some point in future aimed to be stable. Surely this is the reason of most peoples complaints today, all of us who are using Asterisk in real world, commercial environments get extremely frustrated when 'key' issues get fixed in the 'head' release, for example, recent fixes for IAX and SIP voice quality, and are not back ported to the stable/release/whatever version. It leaves us in a very difficult position, as commercially we are placing our users at unnecessary risk by using the 'head' version to get a specific bug fix, but also giving them poor service if we stick with the 'broken' version. Please can those responsible have some understanding of this, as a rule would it not make sense that all (or at least all major) 'fixes' go into both after being appropriately tested, and keep 'head' for the more 'bleeding edge' new features and more radical changes etc? Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P HDLC configuration
Hey Vasyl, this doesn't bode well for me I am going to hate having to recompile a new kernel, and zaptel, asterisk, etc, and restart everything This sucks M On Sunday, May 23, 2004, at 12:40 PM, Vasyl Rublyov wrote: Thank you Michael, I used that sethdlc which is in latest zaptel, sethdlc --version does not work, but sethdlc hdlc0 --version works sethdlc --version --version: unable to get interface information: No such device /sbin/sethdlc hdlc0 --version sethdlc version 1.15 Copyright (C) 2000 - 2003 Krzysztof Halasa [EMAIL PROTECTED]> Today, I am going to try downgrade the kernel to 2.4.19, so it will use old HDLC API. Michael Rowley MD FP
RE: [Asterisk-Users] Development SOP - was:Downgrading Asterisk
For those still impacted by the iax2/gsm/cisco choppy sound, please add your comments to bug #1742. The source of the problem tends to be the asterisk box originating the iax2/gsm data flows (eg, if you hear choppy audio, the * box at the distant end is the one originating inconsistent timestamps) so be sure to include * version data for both ends (if possible). I'm willing to open my system up for those developers that cannot duplicate the problem on their own systems. I have a nice flat network, good hardware, no off-the-wall configurations, an up-to-date kernel and server hardware, etc. Contact me on or off list and I'll arrange for SSH access for you, after we have a short phone conversation. Nik Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, May 28, 2004 7:59 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Downgrading Asterisk The code changes that fixed the cisco choppy sound for Stable went in last Friday. That change corrected iax2 issues that had been known for well over a month but never got applied to Stable. That same code is in Head, however many other changes have happened to Head, and some of those apparently have impacted at least some of us (mostly cisco users). Stable has a number of other bugs that reportedly will never get fixed as the fixes use functionality that exists only in Head. It seems the choppy (and almost unusable) audio in Head is only impacting some cisco users, and since these problems are not impacting the few that can read code, use cisco phones, and are impacted, we're stuck with the problem. The problem seems to be very evasive, however switching the iax2 links to use only iLBC (and not gsm) has corrected issues for some. Although many of us that have worked in a production I/T arena assume something called Stable would truly have known bugs fixed, that's hardly the case for *. That branch really should be renamed to something like v1.0 and remove any reference to Stable and bug fixes as its treated as a lockdown for added functionality, and has nothing to do with functional stability. FYI Downgrading to -stable totally fixed the choppy audio on Cisco my 7960 - * - IAX setup. Now, when would a fix that goes into stable get into the current source (HEAD)? And, isn't checking stuff into a stable branch that doesn't exist elsewhere in the source tree break some rules somewhere? It has to. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nik Martin Sent: Tuesday, May 25, 2004 2:53 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Downgrading Asterisk I upgraded to the latest HEAD version of asterisk, and all IAX calls started sounding choppy. It was suggested on the IRC channel that I go back to asterisk -stable to determine if that fixes it. Is downgrading as simple as upgrading? Because now, -stable builds fine, but I get an error on the asterisk console when starting, something about ast_get_txt not found. Recompiling and installing asterisk HEAD afterwards works just fine. As a side note, I recently upgraded my kernel to 2.4.26 and had an issue with old kernel headers, but have since resolved that prior to trying this downgrade. Any ideas? Nik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
