RE: [Asterisk-Users] Problems compiling on mandrake
I have it installed and working 100% on Mandrake 10.1 Maybe missing development libs are the cause. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: 18 February 2005 09:53 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Problems compiling on mandrake Guys.. Im having problems compiling asterisk under Mandrake 10.1 with kernel 2.6.8.1-12mdk The error I get is: In file included from chan_phone.c:36: /usr/include/linux/ixjuser.h:353: error: syntax error before '*' token make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/root/software/asterisk-1.0.0/channels' make: *** [subdirs] Error 1 Any ideas what might be wrong? __ Anton Krall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems compiling on mandrake
I also tried installing rpm and after running asterisk -vvvc I got this error: [codec_ilbc.so] = (iLBC/PCM16 (signed linear) Codec Translator) Ouch ... error while writing audio data: : Broken pipe Warning, flexibel rate not heavily tested! :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nic le Roux Sent: Viernes, 18 de Febrero de 2005 02:15 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Problems compiling on mandrake I have it installed and working 100% on Mandrake 10.1 Maybe missing development libs are the cause. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: 18 February 2005 09:53 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Problems compiling on mandrake Guys.. Im having problems compiling asterisk under Mandrake 10.1 with kernel 2.6.8.1-12mdk The error I get is: In file included from chan_phone.c:36: /usr/include/linux/ixjuser.h:353: error: syntax error before '*' token make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/root/software/asterisk-1.0.0/channels' make: *** [subdirs] Error 1 Any ideas what might be wrong? __ Anton Krall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP!!!!!!!!
Thank you so much, it worked! Yes turn off silence suppression. xlite - Menu - Advanced - audio settings - Silence Settings - transmite Silence: (change to yes) - Original Message - From: Julius Kidubuka To: asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 10:04 AM Subject: [Asterisk-Users] HELP Hi, I have installed two X-Lite phones and they're able to login successfully. The two phones plus the Asterisk system are all on the same LAN with private addresses assigned to each of them. When a call is initiated and is picked up on the other end, there is completely no sound at all (as in the line goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX. From the Asterisk CLI I see the following errors; i) Unknown RTP codec 72 received ii) RFC3389 support incomplete Anyone got ideas on how I can go about this? Thanks in advance. Julius Kidubuka When you do the common things in life in an uncommon way, you will command the attention of the world -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems compiling on mandrake
Seems you have to change something in the xuser or simething file.. Just read it on the wiki, add a , Now it compiled without errors... On to test the demo system :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nic le Roux Sent: Viernes, 18 de Febrero de 2005 02:15 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Problems compiling on mandrake I have it installed and working 100% on Mandrake 10.1 Maybe missing development libs are the cause. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: 18 February 2005 09:53 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Problems compiling on mandrake Guys.. Im having problems compiling asterisk under Mandrake 10.1 with kernel 2.6.8.1-12mdk The error I get is: In file included from chan_phone.c:36: /usr/include/linux/ixjuser.h:353: error: syntax error before '*' token make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/root/software/asterisk-1.0.0/channels' make: *** [subdirs] Error 1 Any ideas what might be wrong? __ Anton Krall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with starting music on hold when cal l connects to phone via queue
I had similar problems, transferring a call from a queue with # transfer did not work too. Solution for me was to update to CVS-HEAD-02/13/05. This fixed a lot other problems too. Hope, this helps... Guido Hecken Von: Senyo Gualt-Williams [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 18. Februar 2005 01:16 An: Asterisk-Users@lists.digium.com Betreff: [Asterisk-Users] Problem with starting music on hold when call connects to phone via queue Help me Obi-wan, you are my only hope. I am running into an issue where when calls are connected to phones via the queue, when I place the call on hold, no music on hold starts. I am looking at what asterisk is doing with asterisk -r and see no attempt by asterisk to start the MOH. Has anyone else encountered this? I have set all the proper configurations in queues.conf and agents.conf for the MOH, the output from asterisk leads me to believe that these would have no affect on the issue at hand. Thanks in advance for any help, ~Senyo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A104 - D-Channel problem
On Fri, Feb 18, 2005 at 03:38:28AM +0100, Michael Bielicki wrote: upgrade to the following wanpipe and also upgrade the firmware o the crd (it's included in the wanpipe softwaare) ftp://ftp.sangoma.com/linux/custom/2.3.2/wanpipe-beta5g-2.3.2.tgz I did it before asking on the list. I have firmware ver8 on card and wanpipe-beta5g-2.3.2 but problem still exists. Here is wanpipe1.conf from wancfg [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 10 PCIBUS = 0 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL ACTIVE_CH = ALL TE_HIGHIMPEDANCE= NO INTERFACE = V35 CLOCKING= EXTERNAL BaudRate= 0 MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO [w1g1] PROTOCOL= HDLC HDLC_STREAMING = YES ACTIVE_CH = ALL IDLE_FLAG = 0x7E MTU = 1500 MRU = 1500 TDMV_SPAN = 1 TDMV_ECHO_OFF = NO MULTICAST = NO TRUE_ENCODING_TYPE = NO ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] functional difference: canreinvite=yes, no, or update
[EMAIL PROTECTED] wrote: Kevin P. Fleming wrote: I have a patch in my local system that allows the canreinvite setting (which I renamed) to actually be based on IP address masking, so that Asterisk can make a more intelligent decision, but even that has problems, because we don't actually _know_ that any given IP route is available. Actually, we could solve Matthew's problem by checking the IP addresses against the localnet setting and checking if both phones are on the same side. If both are within the localnet, we can reinvite. If both are on public side, we can reinvite. But if one is localnet and one is public, we could automagitically disable reinvite. This should really be the default behaviour if canreinvite=yes and localnet is set to something. Hmmm. Time to code. This is very common problem we come accross as well. I guess coding will make quite a lot people happy... :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Timing device OpenBSD
Hi all, I've been searching the wiki and google for a couple of days now but cannot find any reference to a timing source on OpenBSD. I have * CVS-v1-0-02/15/05-21:54:52 (I always do a cvs -q up -Pd before compiling) running like a charm on OpenBSD 3.6 Now I want to setup some IAX trunks to work and 3 friends and some meetme rooms but it looks like I need a zaptel timing source. Anyone can point me in the right direction ? Thanks -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A104 - D-Channel problem
While on sangoma We are getting a samngom pri?Is there any driver I need to install,how does it work,like a Zaptel card. Any doc Please Let me know altus On Fri, 2005-02-18 at 11:06, Kumak wrote: On Fri, Feb 18, 2005 at 03:38:28AM +0100, Michael Bielicki wrote: upgrade to the following wanpipe and also upgrade the firmware o the crd (it's included in the wanpipe softwaare) ftp://ftp.sangoma.com/linux/custom/2.3.2/wanpipe-beta5g-2.3.2.tgz I did it before asking on the list. I have firmware ver8 on card and wanpipe-beta5g-2.3.2 but problem still exists. Here is wanpipe1.conf from wancfg [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 10 PCIBUS = 0 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL ACTIVE_CH = ALL TE_HIGHIMPEDANCE= NO INTERFACE = V35 CLOCKING= EXTERNAL BaudRate= 0 MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO [w1g1] PROTOCOL= HDLC HDLC_STREAMING = YES ACTIVE_CH = ALL IDLE_FLAG = 0x7E MTU = 1500 MRU = 1500 TDMV_SPAN = 1 TDMV_ECHO_OFF = NO MULTICAST = NO TRUE_ENCODING_TYPE = NO ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with starting music on hold when call connects to phone via queue
On Thu, Feb 17, 2005 at 04:15:55PM -0800, Senyo Gualt-Williams wrote: start the MOH. Has anyone else encountered this? yes exactly the same problem here. I already posted this a while ago but without getting any response. Would be really nice if we could fix this. Stefan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callback agents cannot transfer calls
I've got the same problem. It works fine on some of my older asterisk boxes that haven't been upgraded, CVS-HEAD-12/09/04, but not on the latest box, CVS-HEAD-01/19/05. I've tried both t, T, and tT no luck, I checked my features.conf and it has [featuremap] blindxfer = # i do not have canreinvite set to yes anywhere in any of my configs it works when its not being made through agentcallback -Ryan David Trcka wrote: Hi, my situation is: incoming call goes into the queue and is picked up by callback agent. The agent then wants to transfer the call to another device (another SIP phone). But 'transfer' button doesn't work and '#' button attempts to start channel monitor. Tried with both Queue(testq) and Queue(testq,tT). Is it meant as a feature that agents won't transfer calls at all? I'll appreciate any help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vonage, broadvoice et al
Hi all, I'm just wondering about these VoIP services -- do you have to sign up one account -per- client that will be using the service? I've got multiple extensions behind my Asterisk box, and I want to be able to allow all my staff to place calls via the provider. So if I sign up for one account, will multiple users behind my Asterisk box be able to make calls, using that same account, at the same time? Or do these providers typically only allow one call to be in place at any point in time? Thanks in advance. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Performance in comparission of SER
How much can be the load (How much register and
calls Asterisk can Handle simultaneously by asterisk) and what will be the
performance of Asterisk (Call Quality) if all the users are on SIP only and uses
same Codec, I have all three codecs loaded G.711, G.723, G.729) without media
support i.e. ("canreinvite=yes"),
Thanks RegardsRitesh
Jalan
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RE: [Asterisk-Users] callback agents cannot transfer calls
I'm not shure, but I think something changed in CVS HEAD concerning the # transfers... In older CVS from December 2004 the # transfer had to be terminated by # to start transferring. In actual CVS, you have to press # twice, type in the number, wait 2 seconds and the call get's transferred. Is this normal behaviour? Guido Hecken -Ursprüngliche Nachricht- Von: Ryan Stark [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 18. Februar 2005 11:03 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] callback agents cannot transfer calls I've got the same problem. It works fine on some of my older asterisk boxes that haven't been upgraded, CVS-HEAD-12/09/04, but not on the latest box, CVS-HEAD-01/19/05. I've tried both t, T, and tT no luck, I checked my features.conf and it has [featuremap] blindxfer = # i do not have canreinvite set to yes anywhere in any of my configs it works when its not being made through agentcallback -Ryan David Trcka wrote: Hi, my situation is: incoming call goes into the queue and is picked up by callback agent. The agent then wants to transfer the call to another device (another SIP phone). But 'transfer' button doesn't work and '#' button attempts to start channel monitor. Tried with both Queue(testq) and Queue(testq,tT). Is it meant as a feature that agents won't transfer calls at all? I'll appreciate any help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MultiLine Sip Phones
Sorry Newbie asking everyones option. I am setting up a couple of small asterisk phone systems for my work, I started using some snom 190 and bt102 sip phones (the bt102 works really well with iLBC), but the complaint from my workmates is there is no way to see if other people are on there phone or not, or what lines are being used. The snom 190 only has 5 function keys, the snom 220 seems a bit over the top for simple users. What suggestions do people have on some sip phones that support multiple (6 or more but 10 or more would be better) keys where I can program extension numbers and lines to and use hint from my asterisk box to give updates out (I assume that's what it is for). I was looking at the 3Com Business Phone 3102 as its not really that expensive and looks like it comes with 18 programmable buttons which is great, has anyone had any experience with these phones and doing this or have any better ideas or suggestions? As an extra note I am in Australia so not all brands are available down here. James Bean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is this a bug or by design? Workaround?
Hi,
I need to use the trailing 5 digits of a callerid. callerid may be anything
from a length of 4 to 10 digits in this case.
Using this:
---
SubString,cid=${CALLERIDNUM}|-5|5
Works great, BUT shows this message:
The use of Substring application is deprecated. Please use ${variable:a:b}
instead
So, I try
-
SetVar(cid=${CALLERIDNUM:-5:5})
The result is a empty string if CALLERIDNUM is less than 5 digits long,
which is NOT the case of SubString. SubString command returns what remains of
the variable,
that is - if CALLERIDNUM is 4 digits in length, it returns 4 digits. If
CALLERIDNUM is 6 digits,
it returns 5 digits.
If this approach should replace Substring - it should behave identically,
shouldn't it?
If by design, is there a workaround?
/Stig
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Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?
quote who=beonice If I understood the little documentation I found on 's', it's supposed to be a catchall for ALL incoming calls. That's why I assumed it would catch a DID as well. If that's not the case, it really should be updated in some meta-doc somewhere. :) s is the start extension if there is not one already provided. When DID comes in, the channel is kindof predialed. This is with most digital calls, SIP, IAX, H323, ISDN/PRI... So what happens if the DID is _not_ a US DID? I've seen users here from Europe and Asia as well ... does each country need its own mapping to catch the appropriate incoming DID? DID's are specific to your system. If you have 4 digit extensions and I was setup as a user, then I would need to send you the 4 digit extension I am trying to get to. When you purchase DID from a provider, be it VoicePulse over IAX2 or your local carrier via a PRI, they will dictate what the DID looks like. Some will be the last 4 digits, others will be all 10. (assuming US). They do this, because it would be to difficult to maintain your extension mapping on their side. You purchase a DID. When a call comes in it says, This is the number they were calling, you do your own matching to whatever extension you want. Now, what about the folks who are trying to call other countries, and potentially be called by other DIDs themselves? I'm assuming this sort of thing is very likely. Usually you do not use wildcards for DIDs. This is because people normally purchase more than one. So, you need to distinguish between phone numbers. I currently have two numbers from VoicePulse, so my extensions.conf has this: (numbers are changed to avoid crank calls) [DID] exten = _4157611829,1,Goto(PublicExtensions,8001,1) exten = _4157611763,1,Goto(PublicExtensions,8003,1) So, all inbound calls from VoicePulse goto this context. I jump from here to the extension I want the external phone number mapped to. If you get multiple numbers (say regional numbers) and you want all of them to goto the same place, you can wildcard like this: (gets past the international numbering differences) [DID] exten = _X.,1,Goto(PublicExtensions,8001,1) -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice Message Matching?
Hi all, is there a way to sense the automated announce messages that are sent by cell phone operators? I would like to switch to my own voicemail system if I dial a coworker's cell ph. number and I am connected to the provider voicemail announce (or if the cellphone is unavailable without voicemail). It's like pattern matching, but with voice. Any way Asterisk could do this? TIA Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Notify PAP2-NA?
On Thu, Feb 17, 2005 at 09:44:49AM +0100, Olle E. Johansson wrote: It is time to check the CVS head (v1.1dev) version of Asterisk now, we are heading towards code freeze and production of a new stable release. We do need help testing all new features, finding bugs, reporting them, fixing them. The new realtime architecture is a major improvement and a good platform for a lot of new future technology in Asterisk. We need it tested and proven before we release version 1.2. Thank you for your support in creating a new version of Asterisk -the Open Source PBX! Awesome, I hope the new jitter buffer makes it in too.. -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2: Connection rejected
On Wed, Feb 16, 2005 at 04:10:17PM -0500, Sergey Kuznetsov wrote: They are the same. That's what I've checked first. Have you restarted Asterisk? Not all changes picked up with a reload, sometimes you have unload/reload the module or do a full restart for all changes to take effect... Hope this helps, Peter Bowyer wrote: On Wed, 16 Feb 2005 15:40:19 -0500, Sergey Kuznetsov [EMAIL PROTECTED] wrote: Hi there, I am having a problem. It looks like this: Feb 16 15:01:10 WARNING[11122]: chan_iax2.c:5546 socket_read: Call rejected by XXX.XXX.XXX.XXX: No authority found -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?
On Thu, Feb 17, 2005 at 06:32:52PM -0800, beonice wrote: To answer my own question, at least partially, here is a quote from the Asterisk Configuration chapter in Paul Mahler's book VoIP Telephony With Asterisk: Table 1. Reserved Extension Names -- Character NameUsage - - -- s Start A call that does not have digits associated with it, for example a loopstart analog line, begins at the s extension Interesting. I don't understand it fully, but I'm sure I will if I stare at it long enough. :) I guess it implies that calls coming from DIDs have digits associated with them. Correct. On ISDN lines, E1, T1 and related digital protocols, details such as CallerID, Dialled Number, CLI Presentation, etc are passed as part of the call setup, before there is any discussion of ringing. So Asterisk can go straight into the part of the script that matches. However, on an analog line, you start with ringing and you still know nothing about the call. CallerID comes later and Dialled number is generally never sent at all. So you always start in s. Hope this helps, -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem : undefined symbol.
On Thu, Feb 17, 2005 at 01:05:30PM +0200, Michael Manousos wrote: Did you try asterisk-oh323? http://www.inaccessnetworks.com/projects/asterisk-oh323 Is there any particular reason to prefer oh323 over the builtin h323? I can't find any feature comparison and I can't have both since they require completely different versions of OpenH323. Thanks in advance, -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call termination database
On Thu, Feb 17, 2005 at 09:07:46PM +, Alistair Cunningham wrote: Gonzalo, Yes, pricing would be included, as would minimum call volumes. Providers could choose not to disclose these, but then they'd be shown at the bottom of the page. A feedback system is a good idea; I'll think about how to do it. I was thinking about letting people provide a quality rating, maybe 1-5 on call quality. This would allow people to compare price/quality and aim for where they feel comfortable. But I think it's an awesome idea... -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Amateur - Problema when installing
Friends, I'm in trouble, I tried to install de Asterisk, based on the site manual, into a RedHat 9.0, I followed every step, and it doesn't work. When I does the libpri make install, the message is: quote: [EMAIL PROTECTED] zaptel]# cd .. [EMAIL PROTECTED] src]# cd libpri/ [EMAIL PROTECTED] libpri]# make clean; make install Makefile:93: .depend: No such file or directory ./mkdep -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g `ls *.c` rm -f *.o *.so *.lo *.so.1 *.so.1.0 rm -f testpri testprilib libpri.a libpri.so.1.0 rm -f pritest pridump rm -f .depend Makefile:93: .depend: No such file or directory ./mkdep -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g `ls *.c` cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -c -o pri.o pri.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -c -o q921.o q921.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -c -o prisched.o prisched.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -c -o q931.o q931.c ar rcs libpri.a pri.o q921.o prisched.o q931.o ranlib libpri.a cc -fPIC -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -o pri.lo -c pri.c cc -fPIC -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -o q921.lo -c q921.c cc -fPIC -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -o prisched.lo -c prisched.c cc -fPIC -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -o q931.lo -c q931.c cc -shared -Wl,-soname,libpri.so.1 -o libpri.so.1.0 pri.lo q921.lo prisched.lo q931.lo /sbin/ldconfig -n . ln -sf libpri.so.1 libpri.so mkdir -p /usr/lib mkdir -p /usr/include install -m 644 libpri.h /usr/include install -m 755 libpri.so.1.0 /usr/lib if [ -x /usr/sbin/sestatus ] ( /usr/sbin/sestatus | grep SELinux status: | grep -q enabled); then restorecon -v //lib/libpri.so.1.0; fi ( cd /usr/lib ; ln -sf libpri.so.1 libpri.so ) install -m 644 libpri.a /usr/lib /sbin/ldconfig [EMAIL PROTECTED] libpri]# /quote: I dont know how to solve this problem, anyone can help me ? Im not a Linux expert. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems compiling pridump utility
-lzap means it's looking more libzap, presumably the zaptel library. Have you got it somewhere where the makefile will find it? Hope this helps, On Thu, Feb 17, 2005 at 05:54:42PM -0500, Arlen Raasch wrote: I do 'make pridump' from the libpri source directory and receive the following: # make pridump cc -o pridump pridump.o -L. -lpri -lzap -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g /usr/bin/ld: cannot find -lzap collect2: ld returned 1 exit status make: *** [pridump] Error 1 I am new to all of this, so I am sure I am missing something obvious, any help will be appreciated. I am using libpri verision 1.0.4 with Fedora Core version 2.6.5-1.358. Note: Asterisk and the kernel modules compiled fine, I would just like to try out this utility. Thanks, -Arlen Raasch -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] quadbri and spandsp
Hello, I have bought a targeta quadbri. I want to realize a PBX server to send and to receive fax on lines BRI. I have installed asterisk and the drivers of quadbri (bristuff_0.2.0-RC7b). I downloaded and installed the module spandsp-0.0.2pre10 When i send a fax from the fax machine to asterisk, the application rx_fax saves the fax in a file on the hard disk. The problem comes when I try to send a fax from the PBX with the function tx_fax. The fax machine receives the call but it prints a blank page (The fax says: mistake of communication). Could anyone help me? Thank you. Blas. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Can't Run
i installed Asterisk on linux FC3 box and i was able to make third party calls from voipjet. I then installed an X100P from digitnetworks and i was able to execute modprobe zaptel and modprobe wcfxo and i had to add some lines to the file: 50-udev.rules before i was able to perform ztcfg without errors. I had no problems running Asterisk before i installed X100P, but this is not the case now. Here is part of the compilation right before it stops: [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [res_odbc.so] = (ODBC Resource) asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_odbc.so: undefined symbol: ast_config_load [EMAIL PROTECTED] voicepet-single-x100p]# i recognize the error, but i don't know what to do with it. i appreciate any suggestion __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-H323
On Thu, Feb 17, 2005 at 10:47:28PM -0800, kolo sos wrote: is there any version mismatch or path needed to have a succesful build? i got an error when i done MAKE to the asterisk-oh323. Obviously people have successfully built it, people here use it all the time. Perhaps you can post the actual error you're getting... -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller ID
I have a question: Why is't possible to see Caller ID on the analog phones? If I'm wrong pls tell me how to do to see Caller ID on analog phones. Thank you. mihaid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MultiLine Sip Phones
The SNOM 190 Phones are working quite stable with the hint feature. We have two customers, with at least 12 SNOM 190/SNOM 200 Phones connected with SIP to Asterisk and our hotline is relativ quiet ;-) If the SNOM Phones are within your budget, I think they could be a good choice. Guido Hecken -Ursprüngliche Nachricht- Von: James Bean [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 18. Februar 2005 11:47 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [Asterisk-Users] MultiLine Sip Phones Sorry Newbie asking everyones option. I am setting up a couple of small asterisk phone systems for my work, I started using some snom 190 and bt102 sip phones (the bt102 works really well with iLBC), but the complaint from my workmates is there is no way to see if other people are on there phone or not, or what lines are being used. The snom 190 only has 5 function keys, the snom 220 seems a bit over the top for simple users. What suggestions do people have on some sip phones that support multiple (6 or more but 10 or more would be better) keys where I can program extension numbers and lines to and use hint from my asterisk box to give updates out (I assume that's what it is for). I was looking at the 3Com Business Phone 3102 as its not really that expensive and looks like it comes with 18 programmable buttons which is great, has anyone had any experience with these phones and doing this or have any better ideas or suggestions? As an extra note I am in Australia so not all brands are available down here. James Bean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3Com Business Phone 3102
is there anyone who tested the 3Com Business Phone 3102 with Asterisk? Florian Hecken, Guido schrieb: The SNOM 190 Phones are working quite stable with the hint feature. We have two customers, with at least 12 SNOM 190/SNOM 200 Phones connected with SIP to Asterisk and our hotline is relativ quiet ;-) If the SNOM Phones are within your budget, I think they could be a good choice. Guido Hecken -Ursprüngliche Nachricht- Von: James Bean [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 18. Februar 2005 11:47 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [Asterisk-Users] MultiLine Sip Phones Sorry Newbie asking everyones option. I am setting up a couple of small asterisk phone systems for my work, I started using some snom 190 and bt102 sip phones (the bt102 works really well with iLBC), but the complaint from my workmates is there is no way to see if other people are on there phone or not, or what lines are being used. The snom 190 only has 5 function keys, the snom 220 seems a bit over the top for simple users. What suggestions do people have on some sip phones that support multiple (6 or more but 10 or more would be better) keys where I can program extension numbers and lines to and use hint from my asterisk box to give updates out (I assume that's what it is for). I was looking at the 3Com Business Phone 3102 as its not really that expensive and looks like it comes with 18 programmable buttons which is great, has anyone had any experience with these phones and doing this or have any better ideas or suggestions? As an extra note I am in Australia so not all brands are available down here. James Bean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is this a bug or by design? Workaround?
Stig Andersson wrote:
So, I try
-
SetVar(cid=${CALLERIDNUM:-5:5})
The result is a empty string if CALLERIDNUM is less than 5 digits long,
which is NOT the case of SubString. SubString command returns what remains of
the variable,
that is - if CALLERIDNUM is 4 digits in length, it returns 4 digits. If
CALLERIDNUM is 6 digits,
it returns 5 digits.
If this approach should replace Substring - it should behave identically,
shouldn't it?
If by design, is there a workaround?
This was fixed in cvs head this week, maybe coming up soon in cvs stable.
/Olle
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Re: [Asterisk-Users] quadbri and spandsp
Did you use the caller parameter? Steve Blas wrote: Hello, I have bought a targeta quadbri. I want to realize a PBX server to send and to receive fax on lines BRI. I have installed asterisk and the drivers of quadbri (bristuff_0.2.0-RC7b). I downloaded and installed the module spandsp-0.0.2pre10 When i send a fax from the fax machine to asterisk, the application rx_fax saves the fax in a file on the hard disk. The problem comes when I try to send a fax from the PBX with the function tx_fax. The fax machine receives the call but it prints a blank page (The fax says: mistake of communication). Could anyone help me? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mISDN+w6692pci errors while loading
Hello It is all very confusing due to little information available :) I have a w6692 PCI card, so 1) What ports or modes i can use it? Currently i am plugged into a T0 port, can it be used? And what's the difference from S0? Please point me to some reading full of clues. 2) Due to lack of my understanding of the modes i can't seem to get the right protocol and layermask values for w6692pci.ko module at insmod time. There was this (http://lists.digium.com/pipermail/asterisk-users/2004-December/076239.html) discussion, but it is not helpful to me :( . Clues are very welcome. TIA. -- Konrads Smelkovs Applied IT sorcery. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with SER
Hi all, I am trying to configure * to work with SER (Sip Express Router),the configuration that I am trying is as follows. I have 2 windows machines running X-Lite soft phonesThe * registers with the SER,I want a call from one X-Lite to asterisk(after registration) which is to be forwarded to other X-Lite.For now, I am running both SER and * on the same machine (IP:10.232.2.249) with * on port 5061 and SER on 5060 The contents of the sip.conf is as follows [general]port=5061...context=from-sip...register = asterisk:[EMAIL PROTECTED]:5060/12345 [ser]type=friendusername=asterisksecret=passwordhost=10.232.2.249:5060 [12345]type=friendusername=12345host=dynamicdtmfmode=inband In extensions.conf [from-sip] exten = 12345, 1, Dial(SIP/12345)exten = 12345, 2, Hangup The call is not established, error is: "499: not acceptable here"What can be the problem?. Am I missing something in the configuration? Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term'___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN channel bank
Hi, I want to install an Asterisk Box in my Network and work with some IP phones and ISDN phones. Is this configuration is possible : -E1AsteriskE1 or T1---channel bankISDN phones Wich type of channel bank can I use to do this config? Wich type of ISDN phones can I use? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disable Loop Detection
Hello, I've got the following situation: - Asterisk1 - SER -- other world | | --Asterisk2 - In addition i'm doing a sort of vhost on the asterisk machines, so there could be 3 seperate companies using 1 asterisk box. If an asterisk1 user calls out to ser, but ser decides to route the call back to asterisk1 (because the called company/number is on the same machine), the call is canceled by asterisk and setup via a local channel, this is not what i want because i want to generate CDR's from ser, but since the call isn't going thru SER there are no CDR beeing generated for these calls. Is there any way to get asterisk to answer the looped back call? _ MSN Webmessenger doet het altijd en overal http://webmessenger.msn.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie question
Hello! When the oprator transfers calls to internal extensions to unavailable or busy extensions, how can I prevent these calls from going to voicemail, and route them back to the oprator? But other calls, ie internal between extensions, and calls coming in via DID should get voicemail if extensions are busy / unavailable? Any help be appreciated. TIA! Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Q.SIG support in CVS
Hi, I just read thru the changelog.txt of the current CVS version and what catched my eye was the following line: 'Adding Q.SIG switchtype option to chan_zap' . But there is no sample config in zapata.conf for Q.SIG and no 'feature-list'. Does this exist anywhere or has anyone already has experience with * and Q.SIG and wants to share ?? Thanks a lot in advance, best regards, Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to install X100p
Wait a second, whats the problem you having? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk + RedHat9 - Libpri problem
I install the Asterisk into a RedHat9, exactly like manual says, and I'm having the attached error message when try to install libpri. Please, help on it. [EMAIL PROTECTED] zaptel]# cd .. [EMAIL PROTECTED] src]# cd libpri/ [EMAIL PROTECTED] libpri]# make clean; make install Makefile:93: .depend: No such file or directory ./mkdep -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g `ls *.c` rm -f *.o *.so *.lo *.so.1 *.so.1.0 rm -f testpri testprilib libpri.a libpri.so.1.0 rm -f pritest pridump rm -f .depend Makefile:93: .depend: No such file or directory ./mkdep -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g `ls *.c` cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -c -o pri.o pri.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -c -o q921.o q921.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -c -o prisched.o prisched.c cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -c -o q931.o q931.c ar rcs libpri.a pri.o q921.o prisched.o q931.o ranlib libpri.a cc -fPIC -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -o pri.lo -c pri.c cc -fPIC -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -o q921.lo -c q921.c cc -fPIC -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -o prisched.lo -c prisched.c cc -fPIC -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -o q931.lo -c q931.c cc -shared -Wl,-soname,libpri.so.1 -o libpri.so.1.0 pri.lo q921.lo prisched.lo q931.lo /sbin/ldconfig -n . ln -sf libpri.so.1 libpri.so mkdir -p /usr/lib mkdir -p /usr/include install -m 644 libpri.h /usr/include install -m 755 libpri.so.1.0 /usr/lib if [ -x /usr/sbin/sestatus ] ( /usr/sbin/sestatus | grep SELinux status: | grep -q enabled); then restorecon -v //lib/libpri.so.1.0; fi ( cd /usr/lib ; ln -sf libpri.so.1 libpri.so ) install -m 644 libpri.a /usr/lib /sbin/ldconfig [EMAIL PROTECTED] libpri]#___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to install X100p
Never mind, i saw it futher down. I guessing that you've plugged a phone line from your telephone jack to the x100p (on the right side) then if you've loaded zaptel wcfxo the in you dialplan add something like this: exten = 100,1,Dial(Zap/1/any telephone number) with out the quotes. try that. One Love! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage, broadvoice et al
Vonage, to my knowledge, does not let you connect your own SIP device to their service. They provide their own IAD. As for Broadvoice, I know people that have successfully deployed asterisk with many people sharing the same account. - Pedro On Fri, 18 Feb 2005 17:54:38 +0800, el Flynn [EMAIL PROTECTED] wrote: Hi all, I'm just wondering about these VoIP services -- do you have to sign up one account -per- client that will be using the service? I've got multiple extensions behind my Asterisk box, and I want to be able to allow all my staff to place calls via the provider. So if I sign up for one account, will multiple users behind my Asterisk box be able to make calls, using that same account, at the same time? Or do these providers typically only allow one call to be in place at any point in time? Thanks in advance. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk + RedHat9 - Libpri problem
make clean; make install Shouldn't ist be make clean; make; make install ? Hope it helps... Guido Hecken -Ursprüngliche Nachricht- Von: Paulo - Ibest [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 18. Februar 2005 14:29 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [Asterisk-Users] Asterisk + RedHat9 - Libpri problem I install the Asterisk into a RedHat9, exactly like manual says, and I'm having the attached error message when try to install libpri. Please, help on it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wrapuptime + agents.conf
hello list, i have problem when i am useing wrapuptime with agents.conf my agents.conf looks like this [agents] autologoff=15 musiconhold = default wrapuptime=5 group=1 agent = 1001,4321,Mark Spencer recordagentcalls=yes my aim is every call needs have wrapuptime of 5000 ms but when ever a call comes its directly connecting not wating any more. your views will be highly regarded with regards Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER/Asterisk consultants in Denver
Keith Burns wrote: Hi, I am looking for SER/Asterisk consultants in Denver, please contact me at [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello Keith, My name is Michael Welter, and I have been installing Asterisk systems for two years. You may call me on 303-718-2804. Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN channel bank
On Fri, 18 Feb 2005, Jeremy SALMON wrote: I want to install an Asterisk Box in my Network and work with some IP phones and ISDN phones. Is this configuration is possible : -E1AsteriskE1 or T1---channel bankISDN phones Wich type of channel bank can I use to do this config? Wich type of ISDN phones can I use? What you need is almost a switch. You can find several manufacturers if you search for isdn bri pri multiplexor Most isdn phones should work. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] finding current codec?
* Should I mail something to digium? ;) fax them the agreement from http://www.digium.com/disclaimer.txt roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiring question for Digium card
Chris Blake wrote: Greetings *`s, I have a Digium TDM01B card which I want to connect to a standard phone socket on the wall, for the purposes of testing [EMAIL PROTECTED] On the 4 pin connector going to the wall socket, I have the wires from a CAT5 cable inserted as follows : Brown/White - Blue/White - Blue - Brown On the 8 pin connector going to the back of the Digium card, how would I position all 8 of the wires ? Get a standard RJ14 cable--the kind with a two-pair (four wire) connector at each end. You can plug the RJ14 into the RJ45 socket on the TDM card. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon 2004 tutorials available?
On Thu, 2005-02-17 at 23:20 -0800, Spencer Nassar wrote: Does anyone know if the tutorial materials from Atricon 2004 are available for download anywhere? I'm particularly interested in Joachim Vanheuverzwijn's Performance and Scalability tutorial slides (Asterisk - building your system for performance and scalability). Thanks! Links to presentations are up at http://www.laimbock.com/asterisk/ Joachim's stuff is at http://www.securax.be/astricon/ Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER/Asterisk consultants in Denver
On Friday 18 February 2005 13:44, Michael Welter wrote: Keith Burns wrote: Hi, I am looking for SER/Asterisk consultants in Denver, please contact me at [EMAIL PROTECTED] [... quoted signature deleted ...] Hello Keith, My name is Michael Welter, and I have been installing Asterisk systems for two years. You may call me on 303-718-2804. You failed the intelligence test. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Weird Echo Problem
Ok I know I'm not the only one having echo problem with asterisk but the weird thing is that when I receive a call from a PSTN line on my TDM04B card I don't have any echo problem at the beginning of the call then after a few minutes I start having echo on my side only (the person calling from a regular phone doesn't have any echo), I think that's the way it usually works... (i.e., only you hear the echo). then it stops and comes back all the way until the call is finished. It does the same thing on outgoing calls from my Cisco 7960 phone to the PSTN line. The echo happens when you make a call from IP to a 2-wire PSTN Phone. I have no problem when it's an internal call from one 7960 to another one. And you shouldn't. I tried a lot of different config in zapata.conf and the one that seems to work the best for now is this one : context=incoming signalling=fxs_ks echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=0 txgain=0 immediate=no busydetect=no callprogress-no musiconhold=default usecallerid=yes callerid=asreceived group=1 channel = 1-8 Any suggestion why it starts doing echo after 5 minutes or so? Martin, I had a similar problem, but the echo would just come and go. What are you using for your FXO ports? Do you have the echo cancel turned on when you create those channels? I'm assuming you do since your FXS ports start at one, and unless you specifically turned it off for your FXO ports, the echo cancel setting (and all other settings) will carry over from the previous channel settings. I have a T100P/PRI for my PSTN connection, and I changed the T100P card to another pci slot on the same motherboard. After that, the echo problem disappeared. I just used echocancel=yes and echotraining=yes. The default for echocancel is 128. You can change that number too. I never thought moving the cards around on the motherboard would matter, but after reading a few other posts and seeing that this fixed some other problems for people, I thought I would try it for my echo problem. In fact, my echo problem started after a support guy at Digium had me move some cards around. I erroneously attributed my new echo to the latest version of Asterisk that he installed and not to the moving of the pci cards. I have also heard that changing motherboards will change things too. But that might introduce different issues Regards, Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Astricon 2004 tutorials available?
Almost all of those links don't work including all of the audio files. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Friday, February 18, 2005 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Astricon 2004 tutorials available? On Thu, 2005-02-17 at 23:20 -0800, Spencer Nassar wrote: Does anyone know if the tutorial materials from Atricon 2004 are available for download anywhere? I'm particularly interested in Joachim Vanheuverzwijn's Performance and Scalability tutorial slides (Asterisk - building your system for performance and scalability). Thanks! Links to presentations are up at http://www.laimbock.com/asterisk/ Joachim's stuff is at http://www.securax.be/astricon/ Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to install X100p
Show channels is only going to show you what channels are
actually in use,
not what is configured. Try 'zap show channels'. If that
indicates zap/1
exists, then your issues are likely in the extensions.conf area.
Post the results of:
zap show channels
relavent part of zapata.conf
relavent part of extensions.conf for both incoming and outgoing calls
desk*CLI zap show channels
No such command 'zap' (type 'help' for help)
Contents of Zapata.conf (as per another suggestion):
context=default
signalling=fxs_ks
echocancel=yes
echotraining=800
echocancelwhenbridged=no
rxgain=3.0
txgain=0.0
immediate=yes
channel = 1
And from extension.conf
[default]
exten = _9.,1,SetCallerID(12345678)
;exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],20,tr)
;exten = _9.,2,Dial(IAX2/${EXTEN:[EMAIL PROTECTED],20,tr)
exten = _9.,2,Dial(Zap/1/${EXTEN:1},20,tr)
exten = _9.,3,Congestion
exten = _9.,4,Busy
exten = _9.,5,Hangup
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 14/02/2005
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Re: [Asterisk-Users] Wiring question for Digium card
All you really need is a two wire/two pin cable with a 6 position modular plug on each end ( incorrectly referenced frequently as an RJxx ) Simply plug into the desired 8 position connectors, and the PSTN wall connection. If you are not US, then it's up to you to find the dialtone from the PSTN I asked Digium for a pinout of the 8 position modular connectors on the TDM card, and never received a response. It would seem that only the two center pins ( 4 5 ) are used, but I can't be sure of that. Digium doesn't seem to know much about this card, as I was told by one support person in no uncertain terms that the FXS module did NOT work in a ground start mode, and have since proven that is not the case. Another user was told that it "should work" The FXS module absolutely can be configured as ground start. What the configuration does at call termination is not yet known. Ground start is an important feature for those of us who are using Asterisk as an interface to our electromechanical switches, interconnecting in a private collectors network. John Novack Michael Welter wrote: Chris Blake wrote: Greetings *`s, I have a Digium TDM01B card which I want to connect to a standard phone socket on the wall, for the purposes of testing [EMAIL PROTECTED] On the 4 pin connector going to the wall socket, I have the wires from a CAT5 cable inserted as follows : Brown/White - Blue/White - Blue - Brown On the 8 pin connector going to the back of the Digium card, how would I position all 8 of the wires ? Get a standard RJ14 cable--the kind with a two-pair (four wire) connector at each end. You can plug the RJ14 into the RJ45 socket on the TDM card. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P and SOHO traditional (analog) telephones
Greetings, I looking for Digium TDM400P... it substitute a complete PABX with 6 lines and 6 extensions for traditional telephnes? Any advice ir link are welcome Thanks in advace Pablo Fernandes ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Astricon 2004 tutorials available?
In article [EMAIL PROTECTED], Patrick [EMAIL PROTECTED] wrote: On Thu, 2005-02-17 at 23:20 -0800, Spencer Nassar wrote: Does anyone know if the tutorial materials from Atricon 2004 are available for download anywhere? I'm particularly interested in Joachim Vanheuverzwijn's Performance and Scalability tutorial slides (Asterisk - building your system for performance and scalability). Thanks! Links to presentations are up at http://www.laimbock.com/asterisk/ Joachim's stuff is at http://www.securax.be/astricon/ The second link doesn't appear to work. :-( Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Weird Echo Problem
Well I have 3 Digium TDM04B (4 port FXO) installed in my server. I use 10 channels out of 12. There's 5 PCI slots on my motherboard, currently they fill the first 3 PCI slots. I can try to move them arround leaving one free PCI slot between each of them. The motherboard I use is a Tyan S2875ANRF with dual opteron, 1GB of RAM, 2x 74GB WD Raptor SATA 10k rpm HD, GeForce 4 MX400 AGP graphic card. I have Fedora Core 3 AMD 64bits installed on it. The motherboard is installed in an Antec 3U Rackmount case. I have a Clipcomm device with 4 FXS ports so it's not part of the server and since I have currently only one user using a wireless analog phone I don't know if I do have echo problem on this phone but I have 30 Cisco 7960 phones working with SIP and my main concern is to have the echo fix on the Cisco phones. I have a lot of stuff in my Rackmount cabinet could it create interference that create the echo? It's a real mess in there for now I have to clean it up but I can't put the entire network down during the week so I'll have to do it in the weekend... Otherwise I'll have a lot of people complaining hehe. Thanks Martin Brian M. Arlinghaus wrote: Ok I know I'm not the only one having echo problem with asterisk but the weird thing is that when I receive a call from a PSTN line on my TDM04B card I don't have any echo problem at the beginning of the call then after a few minutes I start having echo on my side only (the person calling from a regular phone doesn't have any echo), I think that's the way it usually works... (i.e., only you hear the echo). then it stops and comes back all the way until the call is finished. It does the same thing on outgoing calls from my Cisco 7960 phone to the PSTN line. The echo happens when you make a call from IP to a 2-wire PSTN Phone. I have no problem when it's an internal call from one 7960 to another one. And you shouldn't. I tried a lot of different config in zapata.conf and the one that seems to work the best for now is this one : context=incoming signalling=fxs_ks echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=0 txgain=0 immediate=no busydetect=no callprogress-no musiconhold=default usecallerid=yes callerid=asreceived group=1 channel = 1-8 Any suggestion why it starts doing echo after 5 minutes or so? Martin, I had a similar problem, but the echo would just come and go. What are you using for your FXO ports? Do you have the echo cancel turned on when you create those channels? I'm assuming you do since your FXS ports start at one, and unless you specifically turned it off for your FXO ports, the echo cancel setting (and all other settings) will carry over from the previous channel settings. I have a T100P/PRI for my PSTN connection, and I changed the T100P card to another pci slot on the same motherboard. After that, the echo problem disappeared. I just used echocancel=yes and echotraining=yes. The default for echocancel is 128. You can change that number too. I never thought moving the cards around on the motherboard would matter, but after reading a few other posts and seeing that this fixed some other problems for people, I thought I would try it for my echo problem. In fact, my echo problem started after a support guy at Digium had me move some cards around. I erroneously attributed my new echo to the latest version of Asterisk that he installed and not to the moving of the pci cards. I have also heard that changing motherboards will change things too. But that might introduce different issues Regards, Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiring question for Digium card
On Fri, 2005-02-18 at 15:50, Michael Welter wrote: Get a standard RJ14 cable--the kind with a two-pair (four wire) connector at each end. You can plug the RJ14 into the RJ45 socket on the TDM card. Howdy Michael, Thanks for replying...I took your advice and all is working. Whaaa !! Thanks again Regards -- Chris Blake Cell: 082 775 1492 Work: +27 11 782 0840 Fax : +27 11 782 0841 Mail: [EMAIL PROTECTED] Recent research has tended to show that the Abominable No-Man is being replaced by the Prohibitive Procrastinator. -- C.N. Parkinson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage, broadvoice et al {Scanned}
I have two home accounts with Vonage and I allow all the family to use Vonage with there extensions. David On Fri, 2005-02-18 at 08:39 -0500, Pedro wrote: Vonage, to my knowledge, does not let you connect your own SIP device to their service. They provide their own IAD. As for Broadvoice, I know people that have successfully deployed asterisk with many people sharing the same account. - Pedro On Fri, 18 Feb 2005 17:54:38 +0800, el Flynn [EMAIL PROTECTED] wrote: Hi all, I'm just wondering about these VoIP services -- do you have to sign up one account -per- client that will be using the service? I've got multiple extensions behind my Asterisk box, and I want to be able to allow all my staff to place calls via the provider. So if I sign up for one account, will multiple users behind my Asterisk box be able to make calls, using that same account, at the same time? Or do these providers typically only allow one call to be in place at any point in time? Thanks in advance. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Shaw [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax with asterisk
On Feb 17, 2005, at 2:32 PM, Justin Richards wrote: I don't do a lot of faxing, but I would like to know I'm going to receive them when I do get one.. I think therein lies the key to your problem. If you're not doing a lot of faxing then its hard to know if the problem is at your end or if its somewhere else (like your ISP). Sending or receiving a fax every now and then means you can easily fall into this situation: 1) after a successful fax or two during a given period of time you think AHA! this is working without a hitch. 2) after an unsuccessful fax or two during a given period of time you think DANG! this is no longer workingwhat changed at *my* end? At least that has been my experience using several VOIP providers but always the same ISP with no changes at my end. It comes down to the fact that some people are lucky to have an ISP that maintains their network properly while others must suffer at the hands of an ISP that does a really crummy job. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitoring a telco line for MWI through a TDM400 FXO
Folks, I've tried to find a reference, but I've had no luck, and would appreciate your thoughts: I'd like to be able to monitor a telco line for Message Waiting Notification, however I cannot figure out if this capability is available. Detecting either FSK or Stuttered Dial tone would serve, but I can't find anything in the list archives or Wiki with clues as to how this might be achieved. Any advice would be appreciated. Cheers, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 14/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Timing device OpenBSD
IAX trunks require that you have a hardware timing source (from a zaptel interface). I believe you can use the ztdummy driver if you don't have a zaptel interface. Mohit. On Fri, 18 Feb 2005 10:14:51 +0100, Michiel van Baak [EMAIL PROTECTED] wrote: Hi all, I've been searching the wiki and google for a couple of days now but cannot find any reference to a timing source on OpenBSD. I have * CVS-v1-0-02/15/05-21:54:52 (I always do a cvs -q up -Pd before compiling) running like a charm on OpenBSD 3.6 Now I want to setup some IAX trunks to work and 3 friends and some meetme rooms but it looks like I need a zaptel timing source. Anyone can point me in the right direction ? Thanks -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] There are 10 types of people. Those who understand binary, and those who don't. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to install X100p
Did you install the drivers for the x100p (zaptel) first and then install asterisk. and what version of asterisk you using ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Astricon 2004 tutorials available?
On Fri, 2005-02-18 at 14:33 +, Tony Mountifield wrote: [snip] Links to presentations are up at http://www.laimbock.com/asterisk/ Joachim's stuff is at http://www.securax.be/astricon/ The second link doesn't appear to work. :-( Yes just noticed that too. Hadn't visited those link in a while. I am chasing zoa on irc trying to convince him to put the link back up or point me to another location. Thanks for the tip (and from the previous poster too). Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Budgetone 101
Everytime that I make a call to a Budgetone 101 phone. I always see the following:
-- Executing Dial("SIP/1001-bac5", "SIP/1000|20|tT") in new stack -- Called 1000 -- Got SIP response 302 "Moved Temporarily" back from 172.22.5.4 -- SIP/1000-465e is busy
I can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones I can not get to work anymore. Anybody know how to fix this?
Josh
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Re: [Asterisk-Users] any good redhat 9.0 rpm reposiroty?
If its not on rpmfind.net good luck... just goto kernel.org and get the tar-ball. -Matthew - Original Message - From: Muhammad Muzzamil Luqman [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, February 18, 2005 1:48 AM Subject: [Asterisk-Users] any good redhat 9.0 rpm reposiroty? I have been googling for the RPM kernel-source-2.4.25-040218.i386.rpm or kernel-source-2.4.25-040218.i686.rpm for the last 59 hrs and couldn't succeed. Can someone suggest me some good Redhat Linux 9.0 rpm repositories. And will the Debian deb work with redhat or not? Kindest Muhamnmad Muzzamil Luqman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disable Loop Detection
On Fri, 18 Feb 2005, E rikje wrote: Hello, I've got the following situation: - Asterisk1 - SER -- other world | | --Asterisk2 - In addition i'm doing a sort of vhost on the asterisk machines, so there could be 3 seperate companies using 1 asterisk box. If an asterisk1 user calls out to ser, but ser decides to route the call back to asterisk1 (because the called company/number is on the same machine), the call is canceled by asterisk and setup via a local channel, this is not what i want because i want to generate CDR's from ser, but since the call isn't going thru SER there are no CDR beeing generated for these calls. Is there any way to get asterisk to answer the looped back call? I've always heard that SIP can't do looped calls - though I've never thought through why that should be. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to install X100p
On Fri, 2005-02-18 at 14:11 +, Mike Wright wrote: desk*CLI zap show channels No such command 'zap' (type 'help' for help) If that is the case you have no zap loaded. Did you make install in zaptel, then libpri and finally asterisk? -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage, broadvoice et al {Scanned}
With unlimited calling plans you need to read the terms of service. Sharing the account within a household or business usually fits in with that. Reselling services in any way is usually prohibited. Some providers with unlimited plans will allow you to set the outbound caller ID to any number on the account. In cases where they only put unlimited plans on an ATA, you can still connect that to * with an fxo card. Most of the providers have various hunt and multi=ring options that you can configure via their web interface. One thing I have run into with 2 providers is their current inability to have multiple SIP/IAX logins within the same account. If I might want to someday have the California DID's land on a server at the California office, I have to create a new account when I order them. Otherwise, all DID's go to the last server that registered with the provider. It's a nuisance because I have to do the entire new account creation process with the provider for each DID and then wind up with different web interface logins for each. David Shaw wrote: I have two home accounts with Vonage and I allow all the family to use Vonage with there extensions. David On Fri, 2005-02-18 at 08:39 -0500, Pedro wrote: Vonage, to my knowledge, does not let you connect your own SIP device to their service. They provide their own IAD. As for Broadvoice, I know people that have successfully deployed asterisk with many people sharing the same account. - Pedro On Fri, 18 Feb 2005 17:54:38 +0800, el Flynn [EMAIL PROTECTED] wrote: Hi all, I'm just wondering about these VoIP services -- do you have to sign up one account -per- client that will be using the service? I've got multiple extensions behind my Asterisk box, and I want to be able to allow all my staff to place calls via the provider. So if I sign up for one account, will multiple users behind my Asterisk box be able to make calls, using that same account, at the same time? Or do these providers typically only allow one call to be in place at any point in time? Thanks in advance. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Performance in comparission of SER
There is a page about this on the wiki. I've heard from real-world sources that you get about 60-70 G729-PSTN calls on a dual 3.6Ghz Xeon Dell. Since SER doesn't handle the media at all, its theoretical limit is around 5000. -Matthew - Original Message - From: Ritesh Jalan [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, February 18, 2005 4:15 AM Subject: [Asterisk-Users] Asterisk Performance in comparission of SER How much can be the load (How much register and calls Asterisk can Handle simultaneously by asterisk) and what will be the performance of Asterisk (Call Quality) if all the users are on SIP only and uses same Codec, I have all three codecs loaded G.711, G.723, G.729) without media support i.e. (canreinvite=yes), Thanks Regards Ritesh Jalan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More on W6692pci NT mode under chan_misdn
So far i've grasped that to use a card in NT mode it should have layermask=3 as module option. Is it the only thing that sets TE or NT mode for card? Perhaps there are settings in misdn.conf ? I can only get the card to work in TE mode and even then when asterisk is ran as asterisk -vvvgc it exits right after chan_misdn is loaded with theese messages: [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables the asterisk to use hardware which is supported by the new ) debug_init: using stdout for debug log debug_init: using stderr for warning log debug_init: using stderr for error log debug_init: debug_mask = 0 Locking Config Mutex UnLocking Config Mutex Init. Stack on port:1 TE Stack No lower Id port:1 init_stack: Success talkinghead:~ # syslog: Feb 18 16:58:50 talkinghead kernel: MISDN free_device: entitylist not empty in misdn.conf there is [NT cards] context=outgoing ports=1 with ports=1ptp it Segfaults. Clues? -- Konrads Smelkovs Applied IT sorcery. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER/Asterisk consultants in Denver
What part of please contact me at [EMAIL PROTECTED] did you not understand? -Matthew - Original Message - From: Michael Welter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, February 18, 2005 7:44 AM Subject: Re: [Asterisk-Users] SER/Asterisk consultants in Denver Keith Burns wrote: Hi, I am looking for SER/Asterisk consultants in Denver, please contact me at [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello Keith, My name is Michael Welter, and I have been installing Asterisk systems for two years. You may call me on 303-718-2804. Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with config.
Hello all. I am trying to get my second x100p card set up and am having some troubles. My zaptel.conf reads: fxsks=1-2 fxoks=3-4 defaultzone=us loadzone=us before adding this card my zaptel.conf read: fxsks=1 fxoks=2-3 defaultzone=us loadzone=us But now that Ive made the change I am getting the following error when running modprobe wcfxo and of course the same error if I use /sbin/ztcfg The error reads: ZT_CHANCONFIG failed on channel 3: No such device or address (6) Any ideas on how to clean this up? Even though it goes against everything I have found online I even tried fxsks=1,2 because they are two physically different cards. Thanks for your help. Stumped. ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on Solaris 10
Title: Asterisk on Solaris 10 Does anyone have experience compiling Asterisk STABLE 1.0.5 on Solaris 10 for x86? I have looked at http://www.voip-info.org/wiki-Asterisk+Solaris+Support but I'm looking for other people's experience in actually using Asterisk under that platform. We only need SIP and IAX2 channels. Is the STABLE even supposed to compile? The above mentioned page says Since 15/Dec/2004, the CVS HEAD version of asterisk has included support for Solaris. Solaris support is not yet included in the stable releases. What compiler am I supposed to use? Do I need to install gcc? Any help or comments are appreciated. Thank you -Manuel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: quadbri and spandsp
Yes. This is my process: 1.- Create a /tmp/sample.call -- Channel: Zap/G1/X --- Here fax machine number Application: txfax Data: /root/fax.tif -- 2.- Shell in a linux terminal: --- mv /tmp/sample.call /var/spool/asterisk/outgoing/ --- I don't have any 'fax' extension in my extensions.conf Is correct my process? Thank you. Blas. -- Date: Fri, 18 Feb 2005 20:54:01 +0800 From: Steve Underwood [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] quadbri and spandsp To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Did you use the caller parameter? Steve Blas wrote: Hello, I have bought a targeta quadbri. I want to realize a PBX server to send and to receive fax on lines BRI. I have installed asterisk and the drivers of quadbri (bristuff_0.2.0- RC7b). I downloaded and installed the module spandsp-0.0.2pre10 When i send a fax from the fax machine to asterisk, the application rx_fax saves the fax in a file on the hard disk. The problem comes when I try to send a fax from the PBX with the function tx_fax. The fax machine receives the call but it prints a blank page (The fax says: mistake of communication). Could anyone help me? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] This is NUTS!!
G'Day All; So I purchased a Cisco 7960 and am now trying to get it configured for *. No can do without the variuos files/images through a FTPF server. I configured the TFTP server on my RHES 3 box, now to get the required CISCO files. So I contacted CISCO to purchase the required maintenance contract so as to gain access to the download area for the files/images. -WHAT A FRUSTRATION!!- CISCO says, Purchase it from your reseller/dealer. OK. So I call my reseller/dealer and he is having the most difficult time getting this $8.00 product, CON-SNT-CP7960, for me. It is just not worth the time and effort for him. So here I am, a week later, and no CP7960. It looks pretty though!! Can anyone recommend a speedier way to get this CON-SNT-CP7960 from CISCO Thanks Ferg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A bit of a survey: What do do if you need more than 4 C.O. lines
Folks, In light of all the troubles people report when running more than one TDM400 card in a system, I wouldn't mind hearing what your solution of choice would be when having to connect 5 or more analog telco circuits to an Asterisk. I'll try and compile the answers together and get them into the Wiki, as I figure this could be useful knowledge for the community. TIA, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 14/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.5 an MySQL CDR
Is anyone else seeing any problems with CDR when using MySQL, specifically dropped legs of the call? ie: +-+-++-+ | calldate| disposition | lastapp| channel | +-+-++-+ | 2005-02-17 12:44:03 | ANSWERED| Hangup | Zap/2-1 | | 2005-02-17 12:42:03 | ANSWERED| Hangup | Zap/1-1 | | 2005-02-17 12:40:03 | ANSWERED| Hangup | Zap/1-1 | | 2005-02-17 12:38:04 | ANSWERED| Hangup | Zap/2-1 | | 2005-02-17 12:38:03 | ANSWERED| BackGround | Zap/3-1 | | 2005-02-17 12:36:04 | ANSWERED| Hangup | Zap/2-1 | | 2005-02-17 12:36:02 | ANSWERED| BackGround | Zap/3-1 | | 2005-02-17 12:34:03 | ANSWERED| Hangup | Zap/2-1 | +-+-++-+ Each of these calls should contain both a BackGround and Hangup lastapp, yet the first 3 do. I'm seeing this during the day when we're taking more calls, and it seems to be progressive (ie. the longer asterisk is running the worse it gets). At night (even though these calls are continuing to happen, and they're supposed to) I'm not seeing these problems. Any ideas what this could be? Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage, broadvoice et al {Scanned}
Hello! When you say sharing the account do you mean multiple simultaneous outgoing calls or just whoever picks up the phone and get's a dialtone can make the call? -Randy Paul wrote: With unlimited calling plans you need to read the terms of service. Sharing the account within a household or business usually fits in with that. Reselling services in any way is usually prohibited. Some providers with unlimited plans will allow you to set the outbound caller ID to any number on the account. In cases where they only put unlimited plans on an ATA, you can still connect that to * with an fxo card. Most of the providers have various hunt and multi=ring options that you can configure via their web interface. One thing I have run into with 2 providers is their current inability to have multiple SIP/IAX logins within the same account. If I might want to someday have the California DID's land on a server at the California office, I have to create a new account when I order them. Otherwise, all DID's go to the last server that registered with the provider. It's a nuisance because I have to do the entire new account creation process with the provider for each DID and then wind up with different web interface logins for each. David Shaw wrote: I have two home accounts with Vonage and I allow all the family to use Vonage with there extensions. David On Fri, 2005-02-18 at 08:39 -0500, Pedro wrote: Vonage, to my knowledge, does not let you connect your own SIP device to their service. They provide their own IAD. As for Broadvoice, I know people that have successfully deployed asterisk with many people sharing the same account. - Pedro On Fri, 18 Feb 2005 17:54:38 +0800, el Flynn [EMAIL PROTECTED] wrote: Hi all, I'm just wondering about these VoIP services -- do you have to sign up one account -per- client that will be using the service? I've got multiple extensions behind my Asterisk box, and I want to be able to allow all my staff to place calls via the provider. So if I sign up for one account, will multiple users behind my Asterisk box be able to make calls, using that same account, at the same time? Or do these providers typically only allow one call to be in place at any point in time? Thanks in advance. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VONAGE ---- ASTERISK SIP TERMINATION?????
Has anyone out there successfully set up their * box to terminate their VONAGE calls? I (and I am sure lots of others) would love to hear how you did it. Id like to be able to get rid of the extra hardware I have hanging around here and use the ASTERISK machine to handle the SIP termination instead of needing to have a Linksys modem (w/phone) and an additional X100P card. Thanks. Wishing for a solution. ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] API manager - Redirect with ExtraChannel
Hi, We try to do something likesomone did in redirectAPI, but not fully success... This is what we did, Both channel has been setup and talking... Action: RedirectChannel: SIP/210.201.75.100-081b9170ExtraChannel: SIP/route886x-79cbExten:18Context:sipPriority:1 I have two issue: 1. Channel and Extrachannel could be the same tech channel, sip? 2. Always one certain party connected, one disconnect //Zombie ?? why?? Event: LinkChannel1: SIP/210.201.75.100-08168dd0Channel2: SIP/route886x-5550Uniqueid1: 1108739916.2Uniqueid2: 1108739925.3 Action: RedirectChannel: SIP/210.201.75.100-08168dd0ExtraChannel: SIP/route886x-5550Exten: 18Context: sipPriority: 1 Event: NewchannelChannel: AsyncGoto/SIP/route886x-5550State: UpCallerid: unknownUniqueid: 1108739972.4 Event: RenameOldname: SIP/route886x-5550Newname: SIP/route886x-5550MASQUniqueid: 1108739925.3 Event: RenameOldname: AsyncGoto/SIP/route886x-5550Newname: SIP/route886x-5550Uniqueid: 1108739972.4 Event: RenameOldname: SIP/route886x-5550MASQNewname: AsyncGoto/SIP/route886x-5550ZOMBIEUniqueid: 1108739925.3 Event: NewextenChannel: SIP/route886x-5550Context: sipExtension: 18Priority: 1Application: AnswerAppData:Uniqueid: 1108739972.4 Event: NewextenChannel: SIP/route886x-5550Context: sipExtension: 18Priority: 2Application: WaitAppData: 1Uniqueid: 1108739972.4 Response: SuccessMessage: Dual Redirect successful Event: UnlinkChannel1: SIP/210.201.75.100-08168dd0Channel2: AsyncGoto/SIP/route886x-5550ZOMBIEUniqueid1: 1108739916.2Uniqueid2: 1108739925.3 Event: HangupChannel: AsyncGoto/SIP/route886x-5550ZOMBIEUniqueid: 1108739925.3Cause: 16 We use this in the astGUIclient to transfer an active conversation(bothparties) to a meetme room:Action: RedirectChannel: Zap/73-1ExtraChannel: SIP/199testphone-1f3cExten: 8600029Context: defaultPriority: 1where 8600029 is a meetme room.Works very well.Sadly like most obscure features in Asterisk it is not documented anywherevery well. But ExtraChannel in Redirect is the only way to send both partieson a 2-party call into a meetme room so that they can be joined by a 3rdparty(without having a multi-line phone that is).MATT--- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Astricon 2004 tutorials available?
On Fri, 18 Feb 2005 14:33:43 + (UTC), Tony Mountifield Joachim's stuff is at http://www.securax.be/astricon/ The second link doesn't appear to work. :-( You are looking for http://www.astertest.com actually. Joachim has started a new site regarding Asterisk performance testing and forums. -- Leif Madsen http://www.leifmadsen.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Timing device OpenBSD
On 10:00, Fri 18 Feb 05, Mohit Muthanna wrote: IAX trunks require that you have a hardware timing source (from a zaptel interface). I believe you can use the ztdummy driver if you don't have a zaptel interface. Mohit. I see in the readme this needs the Linux kernel sources. As I am running OpenBSD instead of Linux I wonder how I can compile this. As far as I know this is impossible. That's why I was asking for a replacement timing source for OpenBSD. -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID
You should be able to specify your caller ID in your zapata.conf for the port corresponding to your analog phone. I have a question: Why is't possible to see Caller ID on the analog phones? If I'm wrong pls tell me how to do to see Caller ID on analog phones. Thank you. mihaid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Process incoming faxes in Asterisk
Hello All I am looking for a solution that can do this: 1-) Receive incoming fax; 2-) Read content and identify a zone in the fax where there is a hand written name; 3-) Based on name, query a database; 4-) Act based on the result in the database; I understand asterisk can receive fax and redirect it in PDF format. Are there any asterisk users who know if such solution already exist or help where to get it working ? Any help on this is much appreciated ! Best regards, Hakem, This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VAD (Silence suppresion problem)
Hello, I'm trying to use Asterisk as a SIP PBX with H.323 trunk connectivity. Everything works except that calls that comes from the H.323 side do not get audio both ways. Since the other way round works fine (calls to H.323 side), I suspect the problem to be in the way VAD or Silence suppresion is negotiated. Is there a way to disable VAD in the Asterisk for H.323 gatekeeper connectivity ? I have tried with both H.323 and OH323 modules with no success. Regards, Jorge A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to install X100p
Message: 21 Date: Fri, 18 Feb 2005 09:56:42 -0500 From: Giovanni Powell [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Trying to install X100p To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Did you install the drivers for the x100p (zaptel) first and then install asterisk. and what version of asterisk you using Hmm - I actually installed asterisk FIRST - was playing with it then I decided I wanted to try the X100p. So I got the card, installed zaptel and libpri. SO Do I now have to go back and rebuild asterisk? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 14/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] softphone that registers in 2 or more SERs
Hi all Do someone know about a softphone that can register in 2 or more SIP servers? It would be useful for me to have a softphone registered in my company´s SER and in my nacional SIP server. I think X-lite can't do it. Thanks Joao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with starting music on hold when call connects to phone via queue
Thanks for the tip! :) ~Senyo I had similar problems, transferring a call from a queue with # transfer did not work too. Solution for me was to update to CVS-HEAD-02/13/05. This fixed a lot other problems too. Hope, this helps... Guido Hecken Von: Senyo Gualt-Williams [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 18. Februar 2005 01:16 An: Asterisk-Users@lists.digium.com Betreff: [Asterisk-Users] Problem with starting music on hold when call connects to phone via queue Help me Obi-wan, you are my only hope. I am running into an issue where when calls are connected to phones via the queue, when I place the call on hold, no music on hold starts. I am looking at what asterisk is doing with asterisk -r and see no attempt by asterisk to start the MOH. Has anyone else encountered this? I have set all the proper configurations in queues.conf and agents.conf for the MOH, the output from asterisk leads me to believe that these would have no affect on the issue at hand. Thanks in advance for any help, ~Senyo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with starting music on hold when callconnects to phone via queue
On Thu, Feb 17, 2005 at 04:15:55PM -0800, Senyo Gualt-Williams wrote: start the MOH. Has anyone else encountered this? yes exactly the same problem here. I already posted this a while ago but without getting any response. Would be really nice if we could fix this. Stefan I believe we found what the problem is. For some reason it seems like the bridge is not being set correctly when calls come in over the queue. In the version of asterisk we are using (asterisk-1.0.2), chan_sip.c evaluates if (p-owner-bridge) in order to start music on hold, but since the bridge doesn't seem to be set correctly, no on-hold music starts. I'll check to see if this bug has already been reported. Does anyone know if what might be causing this bridge to not be set? If it's just a configuration issue then I would like to avoid raising any bug alarm if it is not needed. Good and bad to hear that I wasn't alone in this issue. Thanks, ~Senyo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk GUI
Hello, I am trying to setup an Asterisk GUI with the help of astman(please visit http://astman.sourceforge.net/am-user-guide.html). I have installed astman and currently assessing my GUI using; http://ipaddress-of-asteriskbox/cgi-perl/am-main.pl I am trying to get the menu options in my GUI to work but to no avail. Currently my parameters are set to; Asterisk Install Directory: /usr/ports/net/asterisk/work/asterisk-1.0.3/ Asterisk Config Directory: /usr/local/etc/asterisk Profile Editor Working Directory: /usr/local/etc/asterisk Any ideas on how I can go about this? Thanks in advance. -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone 101
1.0.5.16 - the latest version. Michael 'Moose' Dinn [EMAIL PROTECTED] 2/18/2005 8:14:41 AM
What firmware are you running on your 101?On Fri, Feb 18, 2005 at 08:04:51AM -0700, Josh Wilson wrote: Everytime that I make a call to a Budgetone 101 phone. I always see the following: -- Executing Dial("SIP/1001-bac5", "SIP/1000|20|tT") in new stack -- Called 1000 -- Got SIP response 302 "Moved Temporarily" back from 172.22.5.4 -- SIP/1000-465e is busy I can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones I can not get to work anymore. Anybody know how to fix this? Josh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] WM Wink timings for Nortel
Does anyone know the default EM Wink timings for Nortel DID ports? The default settings on Asterisk are: ;prewink: Pre-wink time (default 50ms) ;preflash:Pre-flash time (default 50ms) ;wink:Wink time (default 150ms) ;flash: Flash time (default 750ms) ;start: Start time (default 1500ms) ;rxwink: Receiver wink time (default 300ms) ;rxflash: Receiver flashtime (default 1250ms) ;debounce:Debounce timing (default 600ms) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] any good redhat 9.0 rpm reposiroty?
Try this site: http://fedoralegacy.org/ they have most of the things there for RedHat 7.1 on to Fedora Core 1 items. - Original Message - From: Muhammad Muzzamil Luqman [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, February 18, 2005 1:48 AM Subject: [Asterisk-Users] any good redhat 9.0 rpm reposiroty? I have been googling for the RPM kernel-source-2.4.25-040218.i386.rpm or kernel-source-2.4.25-040218.i686.rpm for the last 59 hrs and couldn't succeed. Can someone suggest me some good Redhat Linux 9.0 rpm repositories. And will the Debian deb work with redhat or not? Kindest Muhamnmad Muzzamil Luqman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to install X100p
On 15:40, Fri 18 Feb 05, Mike Wright wrote: Message: 21 Date: Fri, 18 Feb 2005 09:56:42 -0500 From: Giovanni Powell [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Trying to install X100p To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Did you install the drivers for the x100p (zaptel) first and then install asterisk. and what version of asterisk you using Hmm - I actually installed asterisk FIRST - was playing with it then I decided I wanted to try the X100p. So I got the card, installed zaptel and libpri. SO Do I now have to go back and rebuild asterisk? yes -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + RedHat9 - Libpri problem
On Fri, Feb 18, 2005 at 10:29:09AM -0300, Paulo - Ibest wrote: I install the Asterisk into a RedHat9, exactly like manual says, and I'm having the attached error message when try to install libpri. I don't see any errors that should affect it. If you're referring to the Makefile:93: .depend: No such file or directory type errors, just ignore them, they shouldn't be causing any problems. Matthew Fredrickson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: quadbri and spandsp
You need to use the caller parameter. Something like: Channel:Zap/G1/ Application:txfax Data:/root/fax.tif|caller might work better. Regards, Steve Blas wrote: Yes. This is my process: 1.- Create a /tmp/sample.call -- Channel: Zap/G1/X --- Here fax machine number Application: txfax Data: /root/fax.tif -- 2.- Shell in a linux terminal: --- mv /tmp/sample.call /var/spool/asterisk/outgoing/ --- I don't have any 'fax' extension in my extensions.conf Is correct my process? Thank you. Blas. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] functional difference: canreinvite=yes, no, or update
Olle E. Johansson wrote: Actually, we could solve Matthew's problem by checking the IP addresses against the localnet setting and checking if both phones are on the same side. If both are within the localnet, we can reinvite. If both are on public side, we can reinvite. But if one is localnet and one is public, we could automagitically disable reinvite. Yes, that is a start. As long as you are comparing the perceived addresses (which I know you would be, I'm just clarifying for others who are reading this thread), that will work, because it won't matter what private addresses the remote peers may be using behind their NATs. It will still break in bizarre routing scenarios, but people who build those networks are used to dealing with stuff like that. This should really be the default behaviour if canreinvite=yes and localnet is set to something. Agreed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Q.SIG support in CVS
On Fri, Feb 18, 2005 at 02:18:37PM +0100, Kurt Bauer wrote: I just read thru the changelog.txt of the current CVS version and what catched my eye was the following line: 'Adding Q.SIG switchtype option to chan_zap' . But there is no sample config in zapata.conf for Q.SIG and no 'feature-list'. Does this exist anywhere or has anyone already has experience with * and Q.SIG and wants to share ?? Yeah, I've got some experience with it (I'm the one working on it :-) ). Right now we can do send/receive of DivertingLegInformation2 messages, message waiting indication activate/deactivate, and receive of calling name information. Oh, and of coure all your basic PRI stuff, such as call setup and teardown. Matthew Fredrickson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
