Re: [Asterisk-Users] sip log messages every few seconds
On Thursday 08 September 2005 19:05, Martin wrote: > This is a single aastra 9113i sip phone. > > asterisk 1.0.9 > > Why do I keep seeing this in the logs ? > > -- > Sep 8 18:44:25 VERBOSE[18779]: Scheduling destruction of call > '[EMAIL PROTECTED]' in 15000 ms > Sep 8 18:44:31 DEBUG[18779]: Setting NAT on RTP to 0 > Sep 8 18:44:31 VERBOSE[18779]: 11 headers, 2 lines > Sep 8 18:44:31 VERBOSE[18779]: Reliably Transmitting: > NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK7032188a > From: "Unknown" ;tag=as12a1c927 > To: > Contact: > Call-ID: [EMAIL PROTECTED] > CSeq: 102 NOTIFY > User-Agent: Asterisk PBX > Event: message-summary > Content-Type: application/simple-message-summary > Content-Length: 42 Forgot to add...there are no calls going on ___ --Bandwidth and Colocation sponsored by Easynews.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] SIP registration issues
Hello. Is there any know issue with Asterisk 1.0.9 concerning intermittent SIP registration issues. My SIP hard phone (aastra 9133i) and soft phone (xlite) keep losing registration so calls to them go direct to VM although calling to other phones from them works fine. The logs show 'Transmitting (no NAT): SIP/2.0 403 Forbidden' which doesn't occur when they miraculously start working/registering. Asterisk seems to lose the user. Sep 9 11:47:36 VERBOSE[2444]: 12 headers, 0 lines Sep 9 11:47:36 VERBOSE[2444]: Using latest request as basis request Sep 9 11:47:36 VERBOSE[2444]: Sending to 192.168.1.100 : 5060 (non-NAT) Sep 9 11:47:36 VERBOSE[2444]: Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK289a5fe76 From: Martin ;tag=d6d383eca9b6910 To: Martin ;tag=as3c7c47f1 Call-ID: [EMAIL PROTECTED] CSeq: 54943697 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.1.100:5060 Sep 9 11:47:36 NOTICE[2444]: Registration from 'Martin ' failed for '192.168.1.100' Sep 9 11:47:36 VERBOSE[2444]: Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Sep 9 11:47:36 VERBOSE[2444]: Sip read: REGISTER sip:192.168.1.50:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKd88070866 Max-Forwards: 70 Content-Length: 0 To: No User From: No User ;tag=0e8bc4f3c760bc2 Call-ID: [EMAIL PROTECTED] CSeq: 535959059 REGISTER Contact: No User Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 But then, some period of time later, they will start working at random times with no changes. Regards...Martin ___ --Bandwidth and Colocation sponsored by Easynews.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] Firmware upgrade Aastra 480i CT
On Monday 12 September 2005 19:10, Chris Coulthurst wrote: > Does anyone have success in upgrading Aastra/Sayson 480i CT firmware? > > All I get, no matter what I've tried is "Unable to upgrade firmware". > > tftpd is working because the dialplan freshens, and I have aastra.cfg > whatevermacaddressfile.cfg and firmware.st in /tftpboot > > Am I missing something stupid? Is there another way to upgrade it? > > Chris Coulthurst > [EMAIL PROTECTED] Huh. They don't have a tftp server. You have to download it, put it into you tftp server and then offer it. Yes, it's lame. I offered to do it for them but haven't heard anything back. Regards...Martin ___ --Bandwidth and Colocation sponsored by Easynews.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] compiling astrisk
I am trying to compile the astrisk-1.0.9 tarball on a RedHat 9 linux box with dev environment. I get a lot of the following as a result of a make /usr/bin/ld /usr/lib/crtn.o: invalid string offset 1>0 for section `.shstrtab' and final show stopper ./gentone busy 480 620 make[1]:***[busy.h] segmentation fault What do I need to fix in order to get a clean make? ___ --Bandwidth and Colocation sponsored by Easynews.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 asterisk build
I am trying to build asterisk from the 1.0.9 stable tarballs. I'm trying to follow these instructions: http://www.digium.com/downloads/quick_install_zaptel_asterisk.pdf I started with a fresh install of RedHat 9 linux on a box with digium card. I did a rpm -q kernel-sources zlib zlib-devel openssl openssl-devel and verified the requisite packages were installed. I uncompressed and unraveled the zaptel tarball with tar -xzf zaptel-1.0.9.2.tar.gz followed by: cd zaptel-1.9.0.2 make clean;make install This resulted in lots similar to the following: /usr/bin/ld /usr/lib/crtn.o: invalid string offset 1>0 for section `' /usr/bin/ld /usr/lib/crtn.o: invalid string offset 4>0 for section `.shstrtab' with the final show stopper ./gendigits make[1]: *** [tones.h] segmentation fault What do I need to fix in order to get a clean make on these tarballs on a RedHat 9 box? Thanks. Martin ___ --Bandwidth and Colocation sponsored by Easynews.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 asterisk build
From the above I'd look at the RH9 dev environment or memory problems. -- Dave Cotton <[EMAIL PROTECTED]> I did a rpm -q to verify all the requisite packages were installed from http://www.voip-info.org/wiki/index.php?page=Asterisk+RPM I did a minimal RH9 install on a 17Gbyte drive. Can you be more specific on what to look at? Thanks for your help. Martin ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: Vontage Problems
I am a newbie and want to step up to VoIP and switch from analog connetion to my Astrisk/Lineox box. Any suggestions on configuring Vontage and what to get/ask when signing up? Has anyone experienced problems with Vontage and Asterisk. I'm using Asterisk (Current Stable) and Sipura-841 phones.While talking on an outbound call the transmission seems to fade out until the other person can't hear me but I can hear them. I've updated the firmware on the 841 but it had no effect. I've also tested the phones on another server using Teliax for termination and I have not had any trouble. Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: Asterisk-Users Digest, Vol 21, Issue 81
I too had a server room fry and need to replace h/w. So what specific Dell servers did/do you deploy? Where is the link w/Digium/s Dell caveats? I'm using the Digium TDM400 card w/* Date: Thu, 13 Apr 2006 19:19:13 -0500 (CDT) From: Aaron Daniel <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed *shrugs* Ya win some ya lose some. We've spent about 10 grand plus on Digium cards and have been pretty satisfied with ours :) Faxes have been working great for over 6 months and the cards work wonderfully in our Dell servers. They just need more documentation on the different configuration options you can pass on load... I think the only problems we've really had are configuration related, or bad hardware on our part, oh, and a server room fry that took out more than just the Asterisk servers :-P Aaron On Fri, 14 Apr 2006, Tony ROBIN wrote: I am so fed up with Digium cards. My company first owned a TE410P, I installed it in a Dell server and "enjoyed" its instability (we bought it months before Digium warned about the incompatibility issues). Then we switched to a TE411P for the hardware echo cancellation. Now we want to receive fax (< 20/day) on it and guess what ? Since April 2006 (again a few months after we bought our brand new card), "officially, fax communications is not supported with Digium cards" ( http://www.voip-info.org/wiki-Asterisk+fax ). Of course, I should have guessed that it is far too much to ask to a $2495 card ! Is the "fax" extension in Asterisk just there to push us to the competing products ? We hesitated to buy another Digium card after the problems with TE410P, but I told myself it was nice to support Asterisk by buying some Digium cards. Now Digium make us regret our buys and a disappointed customer is a lost customer forever... Too sad... -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pstn bridge to asterisk - phones connected to pstn stop ringing when asterisk answers
- Original Message - From: Gaurav P To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, February 26, 2012 7:00 PM Subject: [asterisk-users] pstn bridge to asterisk - phones connected to pstn stop ringing when asterisk answers Hi All, I am using an Obi110 to bridge my PSTN line to Asterisk. Inbound and outbound calls work fine, but I noticed that phones connected directly to the PSTN line stop ringing as soon as Asterisk answers and rings one of my extensions. I'd like the regular phones to continue ringing so I have the option to use my multi-handset phone that is directly connected to the wall if I'm not close to my VOIP extension. Thanks, Gaurav Im sure your incomming context starts with Answer() command. Remove it. Martin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to stop intruder from registering sip?
When will you people learn ... you set the secret=
and it's one of the many frequent passwords most people sets out of
being lazy ...
that simply says ... guess my password and call through my pbx for free ...
so again ...
1) bad people scan extensions 100-199 and 1000- trying to guess
your password
if you were nice enough to set it within a known statistical easy guess
2) either use complicated passwords and sip accounts other than
100-199 1000- or install the fail2ban
Martin
On Fri, Jun 11, 2010 at 4:55 PM, sean darcy wrote:
> This is a small 12 line system, internal extensions 150 - 180. I didn't
> have a phone on 151. Here's the sip.conf stanza:
>
> ;;[151]
> ;;type=friend
> ;;context=longdistance
> ;;callerid="Conf Room" <151>
> ;;secret=
> ;;host=dynamic
> ;;qualify=yes
> ;;dtmfmode=rfc2833
> ;;allow=all
> ;;defaultuser=151
> ;;nat=yes
> ;;canreinvite=no
>
> There's no DISA. And then somehow (how???) ip address 79.117.17.247
> becomes extension 151 and starts making calls to West Africa.
>
> Now contactdeny and contactpermit over solve the problem. For instance,
> I can't register with my voip provider. I don't care about peers who I
> make calls to, or receive calls from. I'm just stunned someone can
> become a peer and make calls themselves.
>
> How do I fix this in some reasonable way.
>
> sean
>
> [Jun 10 15:51:19] VERBOSE[1662] chan_sip.c: -- Registered SIP '151'
> at 79.117.17.247 port 5060
> [Jun 10 15:51:20] NOTICE[1662] chan_sip.c: Peer '151' is now Reachable.
> (161ms / 2000ms)
> [Jun 10 15:51:20] NOTICE[1662] chan_sip.c: Received SIP subscribe for
> peer without mailbox: 151
> [Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using SIP RTP TOS bits 184
> [Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using SIP RTP CoS mark 5
> [Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using SIP VRTP CoS mark 6
> [Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using UDPTL TOS bits 184
> [Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using UDPTL CoS mark 5
> [Jun 10 15:51:22] VERBOSE[4780] pbx.c: -- Executing
> [01125240212...@longdistance:1] Answer("SIP/151-00ae", "") in new stack
> [Jun 10 15:51:22] VERBOSE[4780] pbx.c: -- Executing
> [01125240212...@longdistance:2] Gosub("SIP/151-00ae",
> "DialOut,s,1(01125240212154
> ,DAHDI/g0)") in new stack
> .
> [Jun 10 15:51:22] VERBOSE[4780] pbx.c: -- Executing [...@dialout:9]
> Dial("SIP/151-00ae", "DAHDI/g0/01125240212154") in new stack
> [Jun 10 15:51:22] VERBOSE[4780] chan_dahdi.c: -- Requested transfer
> capability: 0x00 - SPEECH
> [Jun 10 15:51:22] VERBOSE[4780] app_dial.c: -- Called g0/01125240212154
> [Jun 10 15:51:22] VERBOSE[4780] app_dial.c: -- DAHDI/2-1 is
> proceeding passing it to SIP/151-00ae
> [Jun 10 15:51:23] VERBOSE[4780] app_dial.c: -- DAHDI/2-1 is making
> progress passing it to SIP/151-00ae
> [Jun 10 15:51:23] VERBOSE[4780] app_dial.c: -- DAHDI/2-1 is making
> progress passing it to SIP/151-00ae
> [Jun 10 15:51:25] VERBOSE[4780] app_dial.c: -- SIP/151-00ae
> requested special control 16, passing it to DAHDI/2-1
> [Jun 10 15:51:25] VERBOSE[4780] channel.c: -- Music class default
> requested but no musiconhold loaded.
> [Jun 10 15:51:25] VERBOSE[4780] app_dial.c: -- SIP/151-00ae
> requested special control 20, passing it to DAHDI/2-1
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
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Re: [asterisk-users] How to stop intruder from registering sip?
if you know IP then ban with iptables
iptables -A INPUT -s IP -j REJECT
Martin
On Fri, Jun 11, 2010 at 8:41 PM, Martin wrote:
> When will you people learn ... you set the secret=
> and it's one of the many frequent passwords most people sets out of
> being lazy ...
>
> that simply says ... guess my password and call through my pbx for free ...
>
> so again ...
>
> 1) bad people scan extensions 100-199 and 1000- trying to guess
> your password
> if you were nice enough to set it within a known statistical easy guess
>
> 2) either use complicated passwords and sip accounts other than
> 100-199 1000- or install the fail2ban
>
> Martin
>
> On Fri, Jun 11, 2010 at 4:55 PM, sean darcy wrote:
>> This is a small 12 line system, internal extensions 150 - 180. I didn't
>> have a phone on 151. Here's the sip.conf stanza:
>>
>> ;;[151]
>> ;;type=friend
>> ;;context=longdistance
>> ;;callerid="Conf Room" <151>
>> ;;secret=
>> ;;host=dynamic
>> ;;qualify=yes
>> ;;dtmfmode=rfc2833
>> ;;allow=all
>> ;;defaultuser=151
>> ;;nat=yes
>> ;;canreinvite=no
>>
>> There's no DISA. And then somehow (how???) ip address 79.117.17.247
>> becomes extension 151 and starts making calls to West Africa.
>>
>> Now contactdeny and contactpermit over solve the problem. For instance,
>> I can't register with my voip provider. I don't care about peers who I
>> make calls to, or receive calls from. I'm just stunned someone can
>> become a peer and make calls themselves.
>>
>> How do I fix this in some reasonable way.
>>
>> sean
>>
>> [Jun 10 15:51:19] VERBOSE[1662] chan_sip.c: -- Registered SIP '151'
>> at 79.117.17.247 port 5060
>> [Jun 10 15:51:20] NOTICE[1662] chan_sip.c: Peer '151' is now Reachable.
>> (161ms / 2000ms)
>> [Jun 10 15:51:20] NOTICE[1662] chan_sip.c: Received SIP subscribe for
>> peer without mailbox: 151
>> [Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using SIP RTP TOS bits 184
>> [Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using SIP RTP CoS mark 5
>> [Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using SIP VRTP CoS mark 6
>> [Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using UDPTL TOS bits 184
>> [Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using UDPTL CoS mark 5
>> [Jun 10 15:51:22] VERBOSE[4780] pbx.c: -- Executing
>> [01125240212...@longdistance:1] Answer("SIP/151-00ae", "") in new stack
>> [Jun 10 15:51:22] VERBOSE[4780] pbx.c: -- Executing
>> [01125240212...@longdistance:2] Gosub("SIP/151-00ae",
>> "DialOut,s,1(01125240212154
>> ,DAHDI/g0)") in new stack
>> .
>> [Jun 10 15:51:22] VERBOSE[4780] pbx.c: -- Executing [...@dialout:9]
>> Dial("SIP/151-00ae", "DAHDI/g0/01125240212154") in new stack
>> [Jun 10 15:51:22] VERBOSE[4780] chan_dahdi.c: -- Requested transfer
>> capability: 0x00 - SPEECH
>> [Jun 10 15:51:22] VERBOSE[4780] app_dial.c: -- Called g0/01125240212154
>> [Jun 10 15:51:22] VERBOSE[4780] app_dial.c: -- DAHDI/2-1 is
>> proceeding passing it to SIP/151-00ae
>> [Jun 10 15:51:23] VERBOSE[4780] app_dial.c: -- DAHDI/2-1 is making
>> progress passing it to SIP/151-00ae
>> [Jun 10 15:51:23] VERBOSE[4780] app_dial.c: -- DAHDI/2-1 is making
>> progress passing it to SIP/151-00ae
>> [Jun 10 15:51:25] VERBOSE[4780] app_dial.c: -- SIP/151-00ae
>> requested special control 16, passing it to DAHDI/2-1
>> [Jun 10 15:51:25] VERBOSE[4780] channel.c: -- Music class default
>> requested but no musiconhold loaded.
>> [Jun 10 15:51:25] VERBOSE[4780] app_dial.c: -- SIP/151-00ae
>> requested special control 20, passing it to DAHDI/2-1
>>
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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_
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Re: [asterisk-users] How to stop intruder from registering sip?
lol when then if he knows the IP of his provider plus a few phones he can just allow these ... and problem solved forever Martin On Fri, Jun 11, 2010 at 9:02 PM, Steve Edwards wrote: > On Fri, 11 Jun 2010, Martin wrote: > >> if you know IP then ban with iptables >> >> iptables -A INPUT -s IP -j REJECT > > Ever play http://en.wikipedia.org/wiki/Whac-A-Mole ? > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UDPTL T38 via NAT
>>>>> I've got the following setup : >>>>> [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP] I don't see where your NAT is in this scenario >>>>> Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) >>>>> Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32) >>>>> Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29) >>>>> >>>>> This means my outgoing udptl traffic is correctly translated, but >>>>> somehow i'm sending 172.16.0.156 instead of my public IP address on the >>>>> firewall. What about externip=62.180.xxx.xxx? >>>> Did you try t38pt_usertpsource=yes ? AFAIK this is about a port used for rtp, not ip address... I'm currently trying this over 2 NATs against eachother (yes, the worst case) with some ports forwarded but with rare success. One of those NAT's rewrites a port numbers for some reason (i see the ATA registered on port 50xxx or so, the same for rtp. I think t38pt_usertpsource is meant for such a case...? [asterisk 1.6]-LAN-[NAT gateway]inet-[NAT gateway]-LAN-[ATA]-[FAX] Has anybody some positive experience with this? Any idea why NAT messes up the port numbers? Martin L -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?
Either turn off busydetect or increase the busycount to 5-7 or even more ... (10 should be conservative) busydetect looks for cadence or patterns of the same length ... beep on [X ms] beep off [Y ms] so you can afford to increase busycount and have a few second longer calls / the line is kept longer offhook but you don't get false busy detections Also in US/Canada callprogress will do a better job then busydetect since it looks for specific frequencies of the busy signal and not just noise/beep then silence ... If you're somewhere else then you can hire a coder to tweak callprogress algorithm to your country's busy signal frequencies ... Just record the busy signal with ztmonitor and send to someone for code patch... regards Martin On Wed, Jul 28, 2010 at 4:54 PM, bruce bruce wrote: > Hmmwhat about call waiting? > You mean, when a call comes in on that specific line, it generate two beep > tones and hence the system hangs up thinking it's end of the call? > Interesting!!! > If it is call-waiting do I have to set all of the following off for it to > not give me problem again: > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > canpark=yes > cancallforward=yes > busydetect=yes > busycount=3 > Please elaborate a bit if I am off-topic. > Regards, > Bruce > On Wed, Jul 28, 2010 at 5:38 PM, Danny Nicholas wrote: >> >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce >> Subject: [asterisk-users] Why do Zaptel calls drop all of a sudden? >> Couldbusy detect be the problem? >> >> >> >> I am getting a complain that call on analogue lines (Sangoam A400D) drops >> all of a sudden. Here is what I see in logs: >> >> >> >> Could be callwaiting? >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?
Well for the best test you can call in on that line and fire Echo() app and then you'll see if the lines "hangup by themselves" ... is you use fxsks/fxs_ks signaling and it's supported by your lines then it's that that makes remote hangup possible regards Martin On Fri, Jul 30, 2010 at 9:12 AM, bruce bruce wrote: > Thank Martin, > That makes absolute sense. I have turned busy detect off for now and haven't > heard complains or lines remaining open for a Day. I am in Canada. I just > checked chan_dahdi.conf and I don't see callprogress there at all. So, I > guess the lines are fine for hanging up by themselves. Hope this doesn't > give me probs in future. > Thanks, > Bruce > On Fri, Jul 30, 2010 at 6:18 AM, Martin wrote: >> >> Either turn off busydetect or increase the busycount to 5-7 or even >> more ... (10 should be conservative) >> busydetect looks for cadence or patterns of the same length ... beep >> on [X ms] beep off [Y ms] >> so you can afford to increase busycount and have a few second longer >> calls / the line is kept longer offhook >> but you don't get false busy detections >> >> Also in US/Canada callprogress will do a better job then busydetect >> since it looks for specific frequencies of the busy signal >> and not just noise/beep then silence ... If you're somewhere else then >> you can hire a coder to tweak callprogress algorithm >> to your country's busy signal frequencies ... Just record the busy >> signal with ztmonitor and send to someone for code patch... >> >> regards >> Martin >> >> On Wed, Jul 28, 2010 at 4:54 PM, bruce bruce wrote: >> > Hmmwhat about call waiting? >> > You mean, when a call comes in on that specific line, it generate two >> > beep >> > tones and hence the system hangs up thinking it's end of the call? >> > Interesting!!! >> > If it is call-waiting do I have to set all of the following off for it >> > to >> > not give me problem again: >> > callwaiting=yes >> > usecallingpres=yes >> > callwaitingcallerid=yes >> > threewaycalling=yes >> > transfer=yes >> > canpark=yes >> > cancallforward=yes >> > busydetect=yes >> > busycount=3 >> > Please elaborate a bit if I am off-topic. >> > Regards, >> > Bruce >> > On Wed, Jul 28, 2010 at 5:38 PM, Danny Nicholas >> > wrote: >> >> >> >> From: asterisk-users-boun...@lists.digium.com >> >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce >> >> bruce >> >> Subject: [asterisk-users] Why do Zaptel calls drop all of a sudden? >> >> Couldbusy detect be the problem? >> >> >> >> >> >> >> >> I am getting a complain that call on analogue lines (Sangoam A400D) >> >> drops >> >> all of a sudden. Here is what I see in logs: >> >> >> >> >> >> >> >> Could be callwaiting? >> >> >> >> -- >> >> _ >> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> >> http://www.asterisk.org/hello >> >> >> >> asterisk-users mailing list >> >> To UNSUBSCRIBE or update options visit: >> >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> > >> > -- >> > _ >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > New to Asterisk? Join us for a live introductory webinar every Thurs: >> > http://www.asterisk.org/hello >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detecting fax without Aswer()ing the call first?
Trying to make the fax detection work. My current setup (with no fax) is done
without Answer(), so the call is answered only when someone actually picks-up
the phone. But when the incoming call is fax, I can her the tone and call is
never forwarded to "Fax" extension.
But... Strange thing happens when I (mistakenly) put a call on hold:
-- Executing [youngandson-test@incoming:2]
Gosub("SIP/66.193.176.35-00b8", "process-callerid,s,1") in new stack
-- Executing [s@process-callerid:1] Verbose("SIP/66.193.176.35-00b8",
"3,- Original CallerID: "FREE CALL TOLL" <18009806858> ") in new stack
-- - Original CallerID: "FREE CALL TOLL" <18009806858>
-- Executing [s@process-callerid:2] GotoIf("SIP/66.193.176.35-00b8",
"1?4") in new stack
-- Goto (process-callerid,s,4)
-- Executing [s@process-callerid:4] GotoIf("SIP/66.193.176.35-00b8",
"0?8") in new stack
-- Executing [s@process-callerid:5] GotoIf("SIP/66.193.176.35-00b8",
"0?8") in new stack
-- Executing [s@process-callerid:6] GotoIf("SIP/66.193.176.35-00b8",
"1?7:8") in new stack
-- Goto (process-callerid,s,7)
-- Executing [s@process-callerid:7] Set("SIP/66.193.176.35-00b8",
"CALLERID(num)=18009806858") in new stack
-- Executing [s@process-callerid:8] Return("SIP/66.193.176.35-00b8", "")
in new stack
-- Executing [youngandson-test@incoming:3]
Macro("SIP/66.193.176.35-00b8", "stdexten,210,sip/ra2501") in new stack
-- Executing [s@macro-stdexten:1] Dial("SIP/66.193.176.35-00b8",
"sip/ra2501,360") in new stack
== Using SIP RTP CoS mark 5
-- Called sip/ra2501
-- SIP/ra2501-00b9 is ringing
[2013-02-24 17:05:12] WARNING[6554]: chan_sip.c:8979 process_sdp: ignoring
'video' media offer because port number is zero
-- SIP/ra2501-00b9 answered SIP/66.193.176.35-00b8
-- Locally bridging SIP/66.193.176.35-00b8 and SIP/ra2501-00b9
[2013-02-24 17:05:31] WARNING[6554]: chan_sip.c:8979 process_sdp: ignoring
'video' media offer because port number is zero
[2013-02-24 17:05:31] WARNING[6554]: chan_sip.c:8945 process_sdp: ignoring
'audio' media offer because port number is zero
-- Started music on hold, class 'default', on channel
'SIP/66.193.176.35-00b8'
[2013-02-24 17:05:31] WARNING[6532]: res_musiconhold.c:659 monmp3thread: poll()
failed: Interrupted system call
[2013-02-24 17:05:31] WARNING[6532]: res_musiconhold.c:659 monmp3thread: poll()
failed: Interrupted system call
[2013-02-24 17:05:34] WARNING[6532]: res_musiconhold.c:659 monmp3thread: poll()
failed: Interrupted system call
[2013-02-24 17:05:34] WARNING[6532]: res_musiconhold.c:659 monmp3thread: poll()
failed: Interrupted system call
== Redirecting 'SIP/66.193.176.35-00b8' to fax extension due to CNG
detection
-- Stopped music on hold on SIP/66.193.176.35-00b8
== Spawn extension (incoming, fax, 1) exited non-zero on
'SIP/66.193.176.35-00b8' in macro 'stdexten'
== Spawn extension (incoming, fax, 1) exited non-zero on
'SIP/66.193.176.35-00b8'
-- Executing [fax@incoming:1] Gosub("SIP/66.193.176.35-00b8",
"receive-fax,fax,1") in new stack
-- Executing [fax@receive-fax:1] Verbose("SIP/66.193.176.35-00b8",
"3,Incoming fax from 18009806858") in new stack
-- Incoming fax from 18009806858
-- Executing [fax@receive-fax:2] Set("SIP/66.193.176.35-00b8",
"FAXDEST=/var/spool/fax/incoming") in new stack
-- Executing [fax@receive-fax:3] Set("SIP/66.193.176.35-00b8",
"FAX-FILENAME=20130224-170534 Incoming Fax") in new stack
-- Executing [fax@receive-fax:4] ReceiveFAX("SIP/66.193.176.35-00b8",
"/var/spool/fax/incoming/20130224-170534 Incoming Fax.tif") in new stack
-- Channel 'SIP/66.193.176.35-00b8' receiving FAX
'/var/spool/fax/incoming/20130224-170534 Incoming Fax.tif'
== Using UDPTL CoS mark 5
== Spawn extension (receive-fax, fax, 4) exited non-zero on
'SIP/66.193.176.35-00b8'
Fax is suddenly detected and received! (Im not sure why all these warnings came
up, something misconfigured in music on hold...)
Is there any way to make Asterisk "listen" for CNG tone during the connected
call, eliminating the need for Answer() and Wait()?
Is the fax detection completely impossible when compressed codec (g729, gsm...)
is in use? I've read its unreliable but does not work at all for me.
(Asterisk 1.8.13 installed from Debian repository)
Thanks
Martin
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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
If I need to use SIP, from where to get the suitable firmware for these Cisco IP Phones 7942G? Be careful, not all versions of SIP firmware work with asterisk. I do have 8-3-1 (cmterm-7941_7961-sip.8-3-1)here and it works just fine with my 7961. Downloaded somewhere. Version 9.x is broken, SIP only works over TCP. Where do u download the SIP firmware usually for your Cisco IP Phones? Search for "cmterm-7941_7961-sip.8-3-1.zip" I also have some other files here but I don't remember what was the reason for them :-( Martin Your kindly help is highly appreciated. Regards Bilal I'm using the sip firmware.. It's alright.. I feel like I'm not receiving all the features I should.. But MWI works and multiple call appearance.. --- Tato zpráva neobsahuje viry ani jiný škodlivý kód - avast! Antivirus je aktivní. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Fax detection *11.7
in the sip.conf i specified [general] sendrpid=rpid trustrpid=yes language=de videosupport=yes callevents=yes caninvite=yes There is a typo in the last line above. Should be "canreinvite". AFAIK it's obsoleted in favor of directmedia. BTW, try to set it to NO. BTW, what is the codec order? Fax detection doesn't work reliably over compressed codecs (g729 etc...), in my case didn't work at all... try to add: directmedia=no disallow=all allow=ulaw allow=alaw to your peer definition. Martin --- Tato zpráva neobsahuje viry ani jiný škodlivý kód - avast! Antivirus je aktivní. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] logging all console output?
Yes, you can: asterisk -vvvgcn|tee /tmp/log regards Martin On Thu, 27 Feb 2003, Roy Sigurd Karlsbakk wrote: > hi > > can I log all console output while having console access as with > > asterisk -vvvgc > > ? > -- > Roy Sigurd Karlsbakk, Datavaktmester > ProntoTV AS - http://www.pronto.tv/ > Tel: +47 9801 3356 > > Computers are like air conditioners. > They stop working when you open Windows. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Execute asterisk command in shell
asterisk -r -x 'put the command here' regards Martin On Thu, 27 Feb 2003, Rattana BIV wrote: > Hi, > > Does anyone know how to execute an asterisk command in shell ? > > I wanted to make a script who put extension in asterisk. > > Regards > > Rattana > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Execute asterisk command in shell
asterisk -r -x "this should work" On Thu, 27 Feb 2003, Rattana BIV wrote: > it doesn't work with me =( > > > - Message d'origine - > De : "Martin Pycko" <[EMAIL PROTECTED]> > À : <[EMAIL PROTECTED]> > Envoyé : jeudi 27 février 2003 17:10 > Objet : Re: [Asterisk-Users] Execute asterisk command in shell > > > > asterisk -r -x 'put the command here' > > > > regards > > Martin > > > > On Thu, 27 Feb 2003, Rattana BIV wrote: > > > > > Hi, > > > > > > Does anyone know how to execute an asterisk command in shell ? > > > > > > I wanted to make a script who put extension in asterisk. > > > > > > Regards > > > > > > Rattana > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Collect Digits for CO Blind Transfer
You can also do that using Background application:
[transfer]
exten => s,1,Background,some-file ;it can be silence
exten => _XXX,1,Flash ;collecting the digits
exten => _XXX,2,SendDTMF,${EXTEN}
exten => _XXX,3,Hangup
[called_context]
exten => 1000,1,Goto,transfer|s|1 ;magical number
regards
Martin
On Thu, 27 Feb 2003, Steven Critchfield wrote:
> Not to take away from your prize collection, but for now wouldn't that
> be trivial to do either as a patternmatch with assignment to a variable,
> or slightly less trivialy as a agi app that already has access to the
> getdata function?
>
> On Thu, 2003-02-27 at 17:42, Mark Spencer wrote:
> > Sounds like we need an app_getdata (front end to the "getdata" function).
> >
> > I'll do it for any item in my thinkgeek wishlist (you can search by
> > [EMAIL PROTECTED] or by my name).
> >
> > Mark
> >
> > On Thu, 27 Feb 2003, Ben Clark wrote:
> >
> > > I have a blind transfer feature available to me from my telephone
> > > provider and was wondering if asterisk can take advantage of this so
> > > that when a certain extension is called the user is asked for the 11
> > > digit pstn number they wish to call then asterisk flashes the line,
> > > dials the transfer codes and hangs up. I have figured out how to do
> > > everything except collecting the digits from the user and then using
> > > them in the transfer codes. Is it possible to do this with the
> > > asterisk dial plan?
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> Steven Critchfield <[EMAIL PROTECTED]>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Execute asterisk command in shell
Well do you have asterisk running in the background ? I do "asterisk -r -x help" and it works. regards Martin On Fri, 28 Feb 2003, Rattana BIV wrote: > I lauch in shell : > > asterisk -r -x 'help' > asterisk -r -x "help" > asterisk -r -x help > > Put It just connect to asterisk and provide CLI command line. But the help > doesn't print. > > what do you think ? > > > Regards > Rattana > - Message d'origine - > De : "Martin Pycko" <[EMAIL PROTECTED]> > À : <[EMAIL PROTECTED]> > Envoyé : jeudi 27 février 2003 18:04 > Objet : Re: [Asterisk-Users] Execute asterisk command in shell > > > asterisk -r -x "this should work" > > On Thu, 27 Feb 2003, Rattana BIV wrote: > > > it doesn't work with me =( > > > > > > - Message d'origine - > > De : "Martin Pycko" <[EMAIL PROTECTED]> > > À : <[EMAIL PROTECTED]> > > Envoyé : jeudi 27 février 2003 17:10 > > Objet : Re: [Asterisk-Users] Execute asterisk command in shell > > > > > > > asterisk -r -x 'put the command here' > > > > > > regards > > > Martin > > > > > > On Thu, 27 Feb 2003, Rattana BIV wrote: > > > > > > > Hi, > > > > > > > > Does anyone know how to execute an asterisk command in shell ? > > > > > > > > I wanted to make a script who put extension in asterisk. > > > > > > > > Regards > > > > > > > > Rattana > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk/IVR Newbie
If you want to have a system capable of making one call at a time X100P is a good start. regards Martin On Fri, 28 Feb 2003, Tyrone Mills wrote: > Hello All, > > I'm new to Asterisk/IVR and this list. I know this is the 'users' list, but > the 'dev' list seems dead and this list seems to have a bit of dev related > traffic, as well as some very knowledgeable members. > > I have a hopefully simple question: > > I want to create an IVR system that places outbound calls to deliver a > message to clients. For example, the system places a call to a client and > plays the following message: "Your delivery has arrived, it can be picked up > after 3pm today". > > Something like that. Allow the user to have the message repeated, and then > hangup. I've been looking at the Asterisk Developers Kit, is this what I > should be looking at? > > For what I need, is the X100P more than I need? Not enough? Just right? > > Thanks in advance for your assistance, > > Tyrone > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk/IVR Newbie
The only problem with outgoing call mechanism when used with X100P is that if you're not in the US then 'callprogress' might not work for you and X100P right after you dial the number will assume that the remote side picked up the phone. With analog lines it's dificult to know when the call is being picked up. regards Martin On 28 Feb 2003, Steven Critchfield wrote: > On Fri, 2003-02-28 at 09:31, Tyrone Mills wrote: > > Hello All, > > > > I'm new to Asterisk/IVR and this list. I know this is the 'users' > > list, but the 'dev' list seems dead and this list seems to have a bit > > of dev related traffic, as well as some very knowledgeable members. > > > > I have a hopefully simple question: > > > > I want to create an IVR system that places outbound calls to deliver a > > message to clients. For example, the system places a call to a client > > and plays the following message: "Your delivery has arrived, it can be > > picked up after 3pm today". > > > > Something like that. Allow the user to have the message repeated, and > > then hangup. I've been looking at the Asterisk Developers Kit, is this > > what I should be looking at? > > > > For what I need, is the X100P more than I need? Not enough? Just > > right? > > This was really the best place for this question. This is a asterisk use > question, not developing new features. > > If all you need is 1 outbound call at a time then the X100P could very > well be what you need. It allows you to generate calls to the PSTN. > > What you are requesting looks a lot like the feature of sample.call. > Some outside stimulise creates a file that goes in > /var/spool/asterisk/outgoing and asterisk sees this file and attempts to > dial the line specified. At the point of answer asterisk transfers the > call to a predetermined extension. This extension could be either a > generic message with company name and reason for calling, or be passed > into AGI to also look up information to either be pieced together from > recorded samples, or use festival to make recordings on the fly. > > All of this can be accomplished on a X100P. If you need larger capacity, > then you can think of adding more X100Ps, or start looking digital. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] error in tor2
Use modprobe instead of insmod. If you use insmod then you have to first insmod zaptel. regards Martin On Fri, 28 Feb 2003, Victor Sanchez wrote: > i have error in install module of tor2. > > but it compile good. > > > what happen ? > > ivr2:/usr/src/zaptel # make clean; make install > rm -f torisatool makefw tor2fw.h > rm -f zttool > rm -f *.o ztcfg tzdriver sethdlc > rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo > rm -f gendigits tones.h > rm -f libtonezone* > rm -f tor2ee > rm -f core > cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o gendigits.o > gendigits.c > cc -o gendigits gendigits.o -lm > ./gendigits > gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB > -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype > s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I > /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include > usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP > -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN > DALONE_ZAPATA -c zaptel.c > cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATAmakefw.c -o makefw > ./makefw tormenta2.rbt tor2fw > tor2fw.h > Loaded 69900 bytes from file > gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB > -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype > s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I > /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include > usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP > -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN > DALONE_ZAPATA -c tor2.c > gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB > -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype > s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I > /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include > usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP > -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN > DALONE_ZAPATA -c torisa.c > gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB > -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype > s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I > /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include > usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP > -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN > DALONE_ZAPATA -c wcusb.c > gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB > -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype > s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I > /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include > usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP > -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN > DALONE_ZAPATA -c wcfxo.c > gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB > -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype > s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I > /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include > usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP > -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN > DALONE_ZAPATA -c wcfxs.c > gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB > -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype > s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I > /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include > usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP > -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN > DALONE_ZAPATA -c ztdynamic.c > gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB > -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype > s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I > /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include > usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP > -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN > DALONE_ZAPATA -c ztd-eth.c > ztd-eth.c: In function `ztdeth_notifier': > ztd-eth.c:128: warning: deprecated use of label at end of compound statement > gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB > -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype > s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I > /usr/src/linux/include -I/usr/src/linux/include/net -DMODV
RE: [Asterisk-Users] Asterisk/IVR Newbie
Well it should work in Canada and US. regards Martin On Fri, 28 Feb 2003, Tyrone Mills wrote: > First off, thanks to all who have taken the time to respond! > > I would be using the system in Canada. Do you think I would run into that > problem here? I was hoping to use analog lines, but if this is the case, > then perhaps digital is the way to go. > > Thanks, > > Tyrone > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Martin Pycko > Sent: February 28, 2003 9:39 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Asterisk/IVR Newbie > > > The only problem with outgoing call mechanism when used > with X100P is that if you're not in the US then > 'callprogress' might not work for you and X100P > right after you dial the number will assume that > the remote side picked up the phone. With analog lines > it's dificult to know when the call is being picked up. > > regards > Martin > > On 28 Feb 2003, Steven Critchfield wrote: > > > On Fri, 2003-02-28 at 09:31, Tyrone Mills wrote: > > > Hello All, > > > > > > I'm new to Asterisk/IVR and this list. I know this is the 'users' > > > list, but the 'dev' list seems dead and this list seems to have a bit > > > of dev related traffic, as well as some very knowledgeable members. > > > > > > I have a hopefully simple question: > > > > > > I want to create an IVR system that places outbound calls to deliver a > > > message to clients. For example, the system places a call to a client > > > and plays the following message: "Your delivery has arrived, it can be > > > picked up after 3pm today". > > > > > > Something like that. Allow the user to have the message repeated, and > > > then hangup. I've been looking at the Asterisk Developers Kit, is this > > > what I should be looking at? > > > > > > For what I need, is the X100P more than I need? Not enough? Just > > > right? > > > > This was really the best place for this question. This is a asterisk use > > question, not developing new features. > > > > If all you need is 1 outbound call at a time then the X100P could very > > well be what you need. It allows you to generate calls to the PSTN. > > > > What you are requesting looks a lot like the feature of sample.call. > > Some outside stimulise creates a file that goes in > > /var/spool/asterisk/outgoing and asterisk sees this file and attempts to > > dial the line specified. At the point of answer asterisk transfers the > > call to a predetermined extension. This extension could be either a > > generic message with company name and reason for calling, or be passed > > into AGI to also look up information to either be pieced together from > > recorded samples, or use festival to make recordings on the fly. > > > > All of this can be accomplished on a X100P. If you need larger capacity, > > then you can think of adding more X100Ps, or start looking digital. > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How could I install the asterisk with embededsystem?
Well you need: /usr/sbin/asterisk /usr/lib/asterisk/modules (but not all of them) /etc/asterisk/ (but not all of them) /etc/zaptel.conf (if you use zaptel devies) /lib/modules/linux_version/misc/zaptel.o (if aplicable) /lib/modules-linux_version/misc/zapte_driver_that_you_need.o (if aplicable) libpri library if you use PRI with T1/E1 I guess some dependable libraries that you'll find out about when you want to load asterisk and all you get is a bunch of errors. Anyways you should be able to track things first. ps. you might want to use 'strip' utility on most binaries to get rid of debug info regards Martin On Sat, 1 Mar 2003 [EMAIL PROTECTED] wrote: > Hi,I want to install the asterisk to my Flash disk without source code. I > wonder if somebody could hint me the files that I needed. > > john > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Question about asterisk
Sure, you just add to [general] section context=default_context_for_everybody_that_is_not_authenticated SIP-PSTN is already there. regards Martin On Sat, 1 Mar 2003, Bill Jordan wrote: > I wanted to know if there was any way to setup an asterisk server as a > PSTN gateway? > > That is, I wanted the asterisk server to accept invites from any sip > client and send them through its T1 or FXO cards. > > So far I've only been able to make asterisk accept invites from users it > knows about. Is there any way to make it blindly accept all sip invites > without any authentication? > > Bill > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mp3 playing distorted, or very slowed down...unintelligible.
Make sure that you have in zapata.conf
musiconhold=random
regards
Martin
On Sun, 2 Mar 2003, Lenny Tropiano / asterisk.org Mailing list wrote:
>
>
> I have the following in extensions.conf:
>
> [global]
> MP3ROOT=/var/lib/asterisk/mohmp3
>
> [default]
> exten => ,1,Answer ; Answer the line
> exten => ,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
> exten => ,3,MP3Player(${MP3ROOT}/sample-hold.mp3)
>
> The command that runs is:
>
> 14030 pts/0S 0:00 /usr/bin/mpg123 -q -s -b 1024 --mono -r 8000
> /var/lib/asterisk/mohmp3/sample-hold.mp3
>
> I hear distorted or very slowed down audio. Unintelligble. (Apparently mpg123 on
> this distribution is really
> mpg321 (replacement)).
>
> I run the command:
>
> /usr/bin/mpg123 -q -s -b 1024 --mono -r 8000
> /var/lib/asterisk/mohmp3/sample-hold.mp3 -w sample.wav
>
> And the wav file plays just fine.
>
> Also I'm having trouble getting the MOH working, I uncommented this line in music:
>
> random => quietmp3:/var/lib/asterisk/mohmp3,-z
>
> And this process is always running:
>
> 13538 pts/0S 0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 -z
> sample-hold.mp3
>
> When I'm put on hold by an extension, I don't hear anything.Am I missing
> something else?
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question
DO you run a recent CVS ? regards Martin On Sun, 2 Mar 2003, Art O'Dea wrote: > I have zaptel and zapata compiled and installed for the X100P in my > system (the * box has the x100p, and the ATA-186 and * are on a class-c > network, all behind a firewall. Calling in to voicemail works fine, but > ringing the ATA gives me the ZT_LOADZONE error, the line continues to > ring when the ATA phone picks up. > > Following your suggestion, I'd guess it would have something to do with > zaptel since ZT_LOADZONE is referenced in zaptel.c - but could it be > something in the ATA-186 configuration? > Art > > > On Fri, 2003-02-28 at 19:51, Art O'Dea wrote: > > > I have an ATA-186 in a SIP configuration (following Shawn Djernes > > > how-to), but I get the following error at the asterisk console when I > > > try to call the phone connected to the ATA: > > > > > > ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device > > > Failed to register zone 'United States / North America': No data > > >available > > > > > > Everything works if I remove indications.conf from /etc/asterisk - > > > actually, I still get an error message, but it doesn't appear to be > > > fatal. > > > > Quick guess, since ZT_* functions have to do with zaptel or zapata, > > make > > sure you compiled and installed them, even if you aren't using zapata > > hardware. > > > > -- > > Steven Critchfield < [EMAIL PROTECTED] > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transcoding
Yes, asterisk does SIP to IAX/H323/PSTN/MGCP etc and any other combination. regards Martin On Sun, 2 Mar 2003, Thomas Jalsovsky wrote: > > Hello, > > Does asterisk do transcoding when the call goes > through the system, codecs are the same but signaling protocol is changed. > example: > SIP with GSM ---> IAX with GSM > > What quality destruction happen when I use transcoding? I know > this is not a concrete/precise question, but I would like to know how is > it in general. > > What CPU performance is needed for transcoding 30 channels e.g. > from GSM to g723.1 in general? > > Thanks in advance, > Thomas > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax support?
let's say you have one T1 span configured like this in zapata.conf context=incoming group = 1 channel => 1-23 then in extensions.conf [incoming] exten => fax,1,Dial,Zap/25 #FXS port that fax is plugged to exten => _X,1,... (the rest) when asterisk detects fax tones on incoming call it's going to look for fax extension in the channel context. If it finds it then the fax call is going to be routed according to the fax extension rule. regards Martin On Mon, 3 Mar 2003, Gene Kochanowsky wrote: > Is there any way to receive and send faxes using a T100 card? If so how is it done? > > Gene Kochanowsky > Solution Sciences, Inc. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice based FAQ and support system?
> I need a system that can: > > - Accept phone calls and give a greeting That's basically playing back a voice file > > - Allow users to select to either leave a message for callback or browse > FAQs IVR menu > > - Present a multi-level voice menu a user can use to drill down to find > their answers in categories of answers IVR menu > > - If they leave a message send an email to let someone know Normal voicemail feature. > > - If they leave a message record as much data about the call, like caller > ID, time, duration and so on Asterisk is configured to deliver that. Look at one of my voicemail messages: Just wanted to let you know you were just left a 0:18 long message (number 1) in mailbox 6161 from "Install Support" <9547097232>, on Wednesday, February 12, 2003 at 08:31:10 AM so you might want to check it when you get a chance. Thanks! > > - Let users pick up their messages by dialing in, or ideally via a web > interface or other Internet based interface. We have a simple www interface to listen to messages. One can also retrieve a message via a certain, defined extension. > > At first, I would need only one phone line, but as the company grows I may > need multiple incoming phone lines. I would like to avoid a T1 as that > would be quite expensive. I would prefer to use just ordinary phone POTS > lines, at least for now. It's always better to use T1. There are a few problems with analog lines. > > Also, I see I could probably use the same system for a company phone > system, as this is what its designed for. Can ordinary phones be plugged > into it somehow? I don't mind buying hardware, as it would be cheaper than Yes, by using a channel bank, digium FXS card or FXS-SIP devices (for example: ATA186) > the answering service! Sure it would :) regards Martin > > Thanks! > > Jim > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callerid
> how to match callerid from 001... ? > and if don't know how many numbers ? You can do it the same way you match extensions: exten => s/_001.,1,blabla exten => s/_00[2-90].,1,bleble regards Martin > > exten => s/0_,Answer don't work- > anything else ? > > tnx > Thomas > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 32 E1 or 64 E1 Configuration ?
But you can connect several asterisk boxes as one system. regards Martin On Tue, 4 Mar 2003, Sphyrna wrote: > NO, THE ASTERISK HAS A PRATICAL LIMIT OF 8 E1S CURRENTLY. THE I/O ERRORS > STOP EVERYTHING > - Original Message - > From: "Florian Overkamp" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, March 04, 2003 8:35 AM > Subject: Re: [Asterisk-Users] 32 E1 or 64 E1 Configuration ? > > > > At 20:16 4-3-2003 +0900, you wrote: > > >Is it possible to support 32 or 64 E1 in a linux box with Wildcard E400P > > >board ? > > >I'd like to make large scale PPS system. > > > > Hmm, 8 to 16 PCI slots ? That'll be a challenge. I think the current > > practical maximum would be 12 or maybe 16 ? > > > > Florian > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How sample.call is proceeded
pbx/pbx_spool.c On Wed, 5 Mar 2003, Rattana BIV wrote: > Hi, > > I wanted to know in which code source the file sample.call is proceeded when > we put it in /var/spool/asterisk/outgoing/ > > I try to make an application to asterisk who check when an user in H323 > (netmeeting) is connect or not. > > Regards > Rattana > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip registration Time
qualify=1000 in sip.conf in the phone config entry regards Martin On Wed, 5 Mar 2003, Mark Spencer wrote: > > But if I close my sip phone and a call goes through it will still wait > > the 25 seconds before it goes to voice mail even though my Sip phone is > > not even on. If I restart Asterisk and do not register my sip phone it > > will go straight to voice mail after no one picks up on Zap/2. Is there > > a way to force asterisk to keep checking on this phone say every few > > seconds (5)? Callers are getting confused as to why the long dead air > > and usually hang up and call back in. > > You can add a "qualify=1000" to your SIP phone and that will cause > Asterisk to consider your SIP phone to be "unavailble" should it take over > 1000ms to respond (1 second). > > Mark > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MSN Messenger Versions
Can you use MSGSM codec with MSN Windows Messenger 4.x ? regards Martin On Thu, 6 Mar 2003, Wade Weppler wrote: > I'll try to be a little more specific: > > http://www.microsoft.com/exchange/downloads/2000/IMClient47.asp > > This one works for me. > > -wade > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk > > Sent: Thursday, March 06, 2003 8:45 AM > > To: [EMAIL PROTECTED]; Jamie Carl > > Subject: Re: [Asterisk-Users] MSN Messenger Versions > > > > microsoft.com > > > > On Thursday 06 March 2003 14:20, Jamie Carl wrote: > > > I seem to have misplaced my copy. Isn't that always the way. > > > > > > Anyone know where I can get an older 4.x version from? > > > > > > Regards, > > > > > > Jamie Carl > > > Email:[EMAIL PROTECTED] > > > PH: +61-414-365-466 > > > > > > > -Original Message- > > > > From: [EMAIL PROTECTED] > > > > [mailto:[EMAIL PROTECTED] Behalf Of William X > > > > Walsh > > > > Sent: Thursday, 6 March 2003 11:50 PM > > > > To: [EMAIL PROTECTED] > > > > Subject: Re: [Asterisk-Users] MSN Messenger Versions > > > > > > > > > > > > > > > > 5 has no SIP support, go back to 4.6 or 4.7 > > > > > > > > On Thu, 2003-03-06 at 04:20, Jamie Carl wrote: > > > > > Hey all, > > > > > > > > > > Just wanted to know what versions of MSN Messenger people have > > > > > > > > working with > > > > > > > > > Asterisk. > > > > > > > > > > I had 4.5 or so working but with the new 5.whatever it seems to > > > > > > > > be a pain in > > > > > > > > > the ass. > > > > > > > > > > Thanks.. > > > > > > > > > > Regards, > > > > > > > > > > Jamie Carl > > > > > Email:[EMAIL PROTECTED] > > > > > PH: +61-414-365-466 > > > > > > > > > > ___ > > > > > Asterisk-Users mailing list > > > > > [EMAIL PROTECTED] > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > -- > > > > William Walsh <[EMAIL PROTECTED]> > > > > Jabber: [EMAIL PROTECTED] > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > Roy Sigurd Karlsbakk, Datavaktmester > > ProntoTV AS - http://www.pronto.tv/ > > Tel: +47 9801 3356 > > > > Computers are like air conditioners. > > They stop working when you open Windows. > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on and on
Do you have tos=lowdelay in iax.conf ? You may also try to turn off the jitterbuffer (jitterbuffer=no). Also make sure that asterisk is really using gsm codec. WHen you do "iax show channels" in the format column it should show number '2' = GSM. Also when you look at Makefiles make sure that optimization is on. regards Martin On Sat, 8 Mar 2003, Krzysztof Bujak wrote: > hmm... > I have adsl to internet and this is the connection I use for tests. > I haven't yet set up QoS as I am testing on clear connection. > And while using for ex. MS Portrait (ueee MS but this is a piece of cool > soft) > for talking with friends via internet I get the quality much better than > that of asterisk, > what is the reason. In both cases there is GSM06.10 codec used. > > Can anyone tell me which codecs I could use (are bulit into) with asterisk > h323 channel? > > Thanks, > BR, > KRiz > - Original Message - > From: "Ron Gage" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Saturday, March 08, 2003 2:57 PM > Subject: Re: [Asterisk-Users] H323 on and on > > > > Hello! > > > > > > On Sat, 2003-03-08 at 10:06, Krzysztof Bujak wrote: > > > Hi all Asterisk Gurus. > > > > > > I am really badly in need of help. Asterisk is very lovely software, but > has one big disadvantage.. > > > lack of documentation.But let's get to the point. > > > > > > 1. Is it normal that I get such a crappy quality with iax, some drops > and clicks? > > > Could anyone with some similar setup check my quality and say if this is > > > what the people are so excited about? ( I used to work as a speech > quality expert for nokia > > > and what I get out of asterisk is far away from the limits I would let > to go out to customers:-( ) > > > > First and foremost, do you have QOS setup on your network and router? > > QOS == Quality of Service, the ability to allow certain data packets to > > be sent quicker through the TCP/IP stack. This is essential to > > maintaining continuous data flow as any interruptions in that dataflow > > will cause "drops and clicks". > > > > -- > > Ron Gage - Saginaw, Michigan > > I am looking for work - resume at http://www.rongage.org/resume.doc > > Electrical Engineering, Linux Programming, Networking > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > === > > Ta wiadomosc zostala sprawdzona na obecnosc wirusow komputerowych przez > system antywirusowy na serwerze IT Form. > > > > > > === > Ta wiadomosc zostala sprawdzona na obecnosc wirusow komputerowych przez system > antywirusowy na serwerze IT Form. > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on and on
but he was asking about iax too: 1. Is it normal that I get such a crappy quality with iax, some drops and clicks? Could anyone with some similar setup check my quality and say if this is what the people are so excited about? ( I used to work as a speech quality expert for nokia and what I get out of asterisk is far away from the limits I would let to go out to customers:-( ) M On 8 Mar 2003, William X Walsh wrote: > > He's using H.323 not iax > > On Sat, 2003-03-08 at 09:58, Martin Pycko wrote: > > Do you have tos=lowdelay in iax.conf ? > > You may also try to turn off the jitterbuffer (jitterbuffer=no). > > Also make sure that asterisk is really using gsm codec. > > WHen you do "iax show channels" in the format column it should > > show number '2' = GSM. > > > > Also when you look at Makefiles make sure that optimization is on. > > > > regards > > Martin > > > > On Sat, 8 Mar 2003, Krzysztof Bujak wrote: > > > > > hmm... > > > I have adsl to internet and this is the connection I use for tests. > > > I haven't yet set up QoS as I am testing on clear connection. > > > And while using for ex. MS Portrait (ueee MS but this is a piece of cool > > > soft) > > > for talking with friends via internet I get the quality much better than > > > that of asterisk, > > > what is the reason. In both cases there is GSM06.10 codec used. > > > > > > Can anyone tell me which codecs I could use (are bulit into) with asterisk > > > h323 channel? > > > > > > Thanks, > > > BR, > > > KRiz > > > - Original Message - > > > From: "Ron Gage" <[EMAIL PROTECTED]> > > > To: <[EMAIL PROTECTED]> > > > Sent: Saturday, March 08, 2003 2:57 PM > > > Subject: Re: [Asterisk-Users] H323 on and on > > > > > > > > > > Hello! > > > > > > > > > > > > On Sat, 2003-03-08 at 10:06, Krzysztof Bujak wrote: > > > > > Hi all Asterisk Gurus. > > > > > > > > > > I am really badly in need of help. Asterisk is very lovely software, but > > > has one big disadvantage.. > > > > > lack of documentation.But let's get to the point. > > > > > > > > > > 1. Is it normal that I get such a crappy quality with iax, some drops > > > and clicks? > > > > > Could anyone with some similar setup check my quality and say if this is > > > > > what the people are so excited about? ( I used to work as a speech > > > quality expert for nokia > > > > > and what I get out of asterisk is far away from the limits I would let > > > to go out to customers:-( ) > > > > > > > > First and foremost, do you have QOS setup on your network and router? > > > > QOS == Quality of Service, the ability to allow certain data packets to > > > > be sent quicker through the TCP/IP stack. This is essential to > > > > maintaining continuous data flow as any interruptions in that dataflow > > > > will cause "drops and clicks". > > > > > > > > -- > > > > Ron Gage - Saginaw, Michigan > > > > I am looking for work - resume at http://www.rongage.org/resume.doc > > > > Electrical Engineering, Linux Programming, Networking > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > === > > > > Ta wiadomosc zostala sprawdzona na obecnosc wirusow komputerowych przez > > > system antywirusowy na serwerze IT Form. > > > > > > > > > > > > > > > > === > > > Ta wiadomosc zostala sprawdzona na obecnosc wirusow komputerowych przez system > > > antywirusowy na serwerze IT Form. > > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > William Walsh <[EMAIL PROTECTED]> > Jabber: [EMAIL PROTECTED] > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on and on
asterisk -vvvcg when the segfault happens gdb ./asterisk core.[pid] regards Martin On Sat, 8 Mar 2003, Ben Clark wrote: > I am also getting a seg fault when asterisk tries to load > chan_oh323.so. What should I try to get it to work? > > [chan_oh323.so] => (OpenH323 Channel Driver) >== Parsing '/etc/asterisk/oh323.conf': Found > Segmentation fault > > > On Saturday, March 8, 2003, at 09:06 AM, Krzysztof Bujak wrote: > > > ...I tried asterisk open h323 but it seg faults:-(... > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Verbose setting changed?
how about "sip debug" ? regards Martin On Sat, 8 Mar 2003, T Aksoy wrote: > Hi, > > On the release of asterisk I was using before this one, I used to issue a "set > verbose 100" command and I would see all the sip registrations taking place. Now > that doesn't seem to work. > > Could someone clarify what value I should use with the "set verbose" command in > order to see sip registrations. > > Thanks > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blacklisting with *80 - What does it do?
It's because you're not using lookupblacklist application on your callflow. regards Martin On Sun, 9 Mar 2003, Jim Archer wrote: > What does blacklisting a call with *80 do? I tried it by dialing from my > cell, which presents caller id. I then blacklisted and the console debug > reported that it did it. But it seemed to have no effect. I could still > call again. > > Thanks... > > Jim > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] variable in extension.conf
you put some definitions in
[globals] section in extensions.conf
later you just use SetGlobalVar variable to change the
values of global variables and then you just take
the value of a variable like this: ${variable} or
like this ${${variable}} or like this ${extension_${EXTEN}} etc.
a trivial example:
[global]
start=1
[local]
exten => s,1,SetGlobalVar,start=1
exten => s,2,Dial,Zap/${start}
is the same as
exten => s,1,Dial,Zap/1
regards
Martin
On Mon, 10 Mar 2003, Rattana BIV wrote:
> Hi,
>
> How can we use Environnement variable in extension.conf ?
>
>
> regards
> rattana
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Call parking - Still haven't solved
put transfer=yes in the begining of zapata.conf after [channels] regards Martin On Mon, 10 Mar 2003, Mike Reiling wrote: > Did that... Doesn't seem to help > > > On Monday, March 10, 2003, at 09:49 AM, James Sharp wrote: > > > > >> parkext => #700; What ext. to dial to park > > > > Try removing the # > > > > > > > > > -- > Mike Reiling > Systems & Network Administrator > SoftCoin, Inc. > 2000 Sierra Point Parkway > Brisbane, CA 94005 > 650-624-3869 - P > 650-624-3899 - F > > It might look like I'm doing nothing, but at the cellular level I'm > really quite busy. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] USB Interfaces
Do you have a proper zaptel.conf and zapata.conf ? When you modprobe do you have anny errors ? What does "ztcfg -vv" says ? regards Martin On Mon, 10 Mar 2003, Brian J. Schrock wrote: > Howdy, > > I just added a second USB converter from Digium and I am having a > problem. When I modprobe the driver for it whichever one gets > discovered first works, but the second one (though looks fine with > zttool) does not work at all. Asterisk directs calls to it etc, but it > never rings and it will not pass dial tone even though asterisk is > seeing it correctly. > Has anyone else had a problem similar to this? > > Brian J. Schrock > Network Engineer, RHCE, CCNA > Anistone Technologies > Phone: 614-537-2817 > FAX: 614-573-7165 > 6926 Avery Rd. > Dublin, OH 43017 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] USB Interfaces
We have some feedback from our customers that sometimes they are able to run two S100U's on a signle machine. regards Martin On Mon, 10 Mar 2003, Ray Dzek wrote: > I was told specifically by Digium that only one USB FXS device was supported > per system. > > > - Original Message - > From: "Brian J. Schrock" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Monday, March 10, 2003 11:30 AM > Subject: [Asterisk-Users] USB Interfaces > > > > Howdy, > > > > I just added a second USB converter from Digium and I am having a > > problem. When I modprobe the driver for it whichever one gets > > discovered first works, but the second one (though looks fine with > > zttool) does not work at all. Asterisk directs calls to it etc, but it > > never rings and it will not pass dial tone even though asterisk is > > seeing it correctly. > > Has anyone else had a problem similar to this? > > > > Brian J. Schrock > > Network Engineer, RHCE, CCNA > > Anistone Technologies > > Phone: 614-537-2817 > > FAX: 614-573-7165 > > 6926 Avery Rd. > > Dublin, OH 43017 > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] USB Interfaces
I think UHCI Martin On Mon, 10 Mar 2003, Brian J. Schrock wrote: > UHCI or OHCI? > > On Monday, March 10, 2003, at 04:23 PM, Martin Pycko wrote: > > > We have some feedback from our customers that sometimes > > they are able to run two S100U's on a signle machine. > > > > regards > > Martin > > > > On Mon, 10 Mar 2003, Ray Dzek wrote: > > > >> I was told specifically by Digium that only one USB FXS device was > >> supported > >> per system. > >> > >> > >> - Original Message - > >> From: "Brian J. Schrock" <[EMAIL PROTECTED]> > >> To: <[EMAIL PROTECTED]> > >> Sent: Monday, March 10, 2003 11:30 AM > >> Subject: [Asterisk-Users] USB Interfaces > >> > >> > >>> Howdy, > >>> > >>> I just added a second USB converter from Digium and I am having a > >>> problem. When I modprobe the driver for it whichever one gets > >>> discovered first works, but the second one (though looks fine with > >>> zttool) does not work at all. Asterisk directs calls to it etc, but > >>> it > >>> never rings and it will not pass dial tone even though asterisk is > >>> seeing it correctly. > >>> Has anyone else had a problem similar to this? > >>> > >>> Brian J. Schrock > >>> Network Engineer, RHCE, CCNA > >>> Anistone Technologies > >>> Phone: 614-537-2817 > >>> FAX: 614-573-7165 > >>> 6926 Avery Rd. > >>> Dublin, OH 43017 > >>> > >>> ___ > >>> Asterisk-Users mailing list > >>> [EMAIL PROTECTED] > >>> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > Brian J. Schrock > Network Engineer, RHCE, CCNA > Anistone Technologies > Phone: 614-537-2817 > FAX: 614-573-7165 > 6926 Avery Rd. > Dublin, OH 43017 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] variable in extension.conf
>From now on (taking about asterisk's CVS) you can access environmental
variables using ${ENV(VARENV)}
regards
Martin
On Tue, 11 Mar 2003, Rattana BIV wrote:
> I try to detect if an user who use Netmeeting is connected or not.
> I think in order to do that, Netmeeting-user open a web page (in PHP) et
> press the button Connect or Disconnect and the PHP set the Environnement
> variable which will be proceeded in extension.conf
>
> So i need Environnement Variable, I have test it with :
> s,1,SetVar,toto=$VARENV where VARENV is my environnement variable but toto
> not take the value. perhaps should I try toto=${VARENV} or toto=${$VARENV}.
>
>
> Regards
> Rattana
>
> - Message d'origine -
> De : "Steven Critchfield" <[EMAIL PROTECTED]>
> À : <[EMAIL PROTECTED]>
> Envoyé : lundi 10 mars 2003 18:27
> Objet : Re: [Asterisk-Users] variable in extension.conf
>
>
> > On Mon, 2003-03-10 at 11:00, Rattana BIV wrote:
> > > Hi,
> > >
> > > How can we use Environnement variable in extension.conf ?
> >
> > I don't think you can at this moment, and I am not sure that would be a
> > good idea as you asterisk thread should be long lived and therefore
> > anything you would put in an environment variable would just as well be
> > placed in a config file. Not to mention a config file can be reread
> > without stopping asterisk or interupting running calls.
> >
> > Could you point out what you wanted to accomplish with environment
> > variables so we can see if there is a better way of communicating the
> > information to asterisk with you, or see what I am currently
> > overlooking.
> > --
> > Steven Critchfield <[EMAIL PROTECTED]>
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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Re: [Asterisk-Users] FIX: iconnect + deltathree
It's received in a SIP header. regards Martin On Tue, 11 Mar 2003, T Aksoy wrote: > Hi Mark, > > Not familiar with "received=". What does it do? Has it got any application > within the nat domain? > > Thanks > Tan > > > - Original Message - > From: "Mark Spencer" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, March 11, 2003 8:25 PM > Subject: [Asterisk-Users] FIX: iconnect + deltathree > > > Okay fellas, thanks to Ravi Sakaria's keen observations, we finally found > what broke. When we added support for ;received=, it broke the ability to > connect to iconnect. CVS Asterisk now only sends the ;received= if nat is > turned on on the connection. Please try it out and get back to me. > > Mark > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Fax Handled: no" config
It's dynamically changed to "Yes" when the fax gets detected on this channel. regards Martin On Wed, 12 Mar 2003, Darrell Eldridge wrote: > I still haven't been able to get fax detection going, > but I came across something: when I execute "zap show > channel 47" one of the parameters shown is "Fax > Handled: no". I assume that's a reflection of > something in zapata.conf, but I don't find anything > there. Should it read "...yes" in order for Asterisk > to detect the fax tones? If so, what's the syntax for > setting it to yes? > > __ > Do you Yahoo!? > Yahoo! Web Hosting - establish your business online > http://webhosting.yahoo.com > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Digits
You may try to add relaxdtmf=yes just before channel => 4 in zapata.conf regards Martin On Wed, 12 Mar 2003, Brian J. Schrock wrote: > I am using background, the pbx-invalid stuff should (if DTMF > recognition is working correctly) not get played. > > On Wednesday, March 12, 2003, at 01:30 PM, Steven Critchfield wrote: > > > Playback is not interuptable, use Background. > > > > On Wed, 2003-03-12 at 12:19, Brian J. Schrock wrote: > >> Hello, > >> > >> I am having a problem with Asterisk that I just cannot get fixed... > >> When I call in to the main number I have to wait until well into the > >> second message shown in the extensions.conf snippet below to enter an > >> extension number. If I enter digits really slowly sometimes it will > >> work during the first message. Usually my callers just get the > >> incredibly annoying "invalid message recording". I remember reading on > >> here about DTMF detection problems earlier, but cannot find anything > >> relevant. Has anyone else had this problem, or does anyone else know > >> what could be the problem? > >> > >> My extension are all 2244,2245,and 2246. If I enter 2244 too early in > >> the playback Asterisk recognizes it as 222 or sometimes 24. > >> > >> If I unload the zap drivers and reload them and restart asterisk it > >> will work just fine for the first call but after that the problem > >> shows > >> up. > >> > >> If I play with txgain and rxgain enough I can make the problem worse > >> but not better. > >> > >> My asterisk is from cvs two days ago. > >> > >> I have been turning echo cancellation on and off in different > >> combinations to see how it affects everything, and it did not have an > >> impact. > >> > >> ##Extensions.conf## > >> > >> [Afternoon] > >> exten => t,1,Goto,default|s|1 > >> exten => i,1,Playback,pbx-invalid > >> exten => i,2,Goto,default|s|1 > >> include => extensions > >> exten => s/_6145551234,1,Answer > >> exten => s/_6145551234,2,Dial,Zap/g2 > >> exten => s,1,Answer > >> exten => s,2,Wait,1 > >> exten => s,3,DigitTimeout,10 > >> exten => s,4,Background,Afternoon_Intro > >> exten => s,5,Background,Exten_Direct > >> > >> ##Zapata.conf## > >> > >> [channels] > >> > >> txgain=0 > >> rxgain=0 > >> context = default > >> language = en > >> callwaiting = yes > >> callwaitingcallerid = no > >> threewaycalling = yes > >> transfer = yes > >> cancelforward = yes > >> callreturn = no > >> usecallerid = yes > >> hidecallerid = no > >> echocancel = yes > >> echocancelwhenbridged = no > >> immediate = yes > >> > >> group = 1 > >> ;use with FXO PCI card > >> signalling = fxs_ks > >> channel => 1-3 > >> > >> echocancel = yes > >> echocancelwhenbridged = yes > >> context = local > >> immediate = no > >> group = 2 > >> txgain=0 > >> rxgain=0 > >> ;use with FXS USB card > >> signalling = fxo_ks > >> callerid = "Brian Schrock" <(614) 798-9106> > >> mailbox=2244,2245,2246 > >> channel => 4 > >> > >> Brian J. Schrock > >> Network Engineer, RHCE, CCNA > >> Anistone Technologies > >> Phone: 614-798-9106 > >> FAX: 614-573-7165 > >> 6926 Avery Rd. > >> Dublin, OH 43017 > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > Brian J. Schrock > Network Engineer, RHCE, CCNA > Anistone Technologies > Phone: 614-798-9106 > FAX: 614-573-7165 > 6926 Avery Rd. > Dublin, OH 43017 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Proposed IAX2 Name
IAX is short and I like it. Besides if that additional '2' irritates you then anyways in the near future when IAX2 is working fine ppl will switch eventually to IAX2 and then we'll refer to IAX2 as IAX Martin On Thu, 13 Mar 2003, Mark Spencer wrote: > What do you all think of renaming IAX2 as: > > Telephony Authentication, Signalling, and Transport Exchange (TASTE) > > "TASTE" is easy to remember and has a sort of ironic relation to "SIP". > Is it took hoaky? > > Mark > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Beginning of voicemail missed by sip phone
You can playback a second or two of silence ... regards Martin On Thu, 13 Mar 2003, T Aksoy wrote: > Hi, > > We are testing a number of sip phones from different manufacturers. With one phone > in particular, when I dial the asterisk voicemail, it misses around half a second > from the beginning of the announcement. I don't have this problem with the snom 200 > or 100. > > Does anyone know why this happens? Is it a sync issue? How do I delay the start of > the voicemail announcement? (Maybe that will fix the problem). > > Thanks > Tan Aksoy > Telappliant Solutions > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Build a complex IVR?
[deeper]
exten => s,1,Playback,you-re-in-the-deepest-menu
exten => s,2,Goto,options|s|1
[options]
exten => s,1,Background,prompt-1-deeper-2-back
exten => 1,1,Goto,deeper|s|1
exten => 2,1,Goto,sales|s|1
[sales]
exten =>
s,1,Background,prompt-1-information-2-connect-or-stay-on-the-line-0-operator-*-to-go-back
exten => s,2,Goto,2
exten => 0,1,Dial,Zap/2
exten => 1,1,PlayBack,information
exten => 2,1,Dial,Zap/1
exten => *,1,Goto,Menu|s|1
[Menu]
exten => s,1,Background,prompt-1-sales-2-options
exten => 1,1,Goto,sales|s|1
exten => 2,1,Goto,options|s|1
exten => i,1,Goto,s
exten => t,1,Goto,s
[incoming]
exten => s,1,Goto,Menu|s|1
regards
Martin
On Thu, 13 Mar 2003, it wrote:
> Hi,Jim,
> Thank you very much for your reply. But I think the menu file you
> designed must be interpreted by another program writen by your but not
> asterisk.
>
> - Original Message -
> From: "Jim Gottlieb" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, March 13, 2003 12:21 AM
> Subject: Re: [Asterisk-Users] Build a complex IVR?
>
>
> > On 2003-03-13 at 15:39, "it" <[EMAIL PROTECTED]> wrote:
> >
> > >I would like to know if Asterisk could be used to build a IVR
> with
> > >complex flow?
> >
> > Yes, we have done this. We use menu files that define what each
> > keypress does so we can build complex menus with full flexibility.
> >
> > A short example (since you asked):
> >
> > default = 0
> > prompt
> > {
> > name = 3046
> > }
> >
> > 0
> > {
> > action = debitcard
> > declines = debitcard-declines
> > pin-prompt = 5122
> > number-prompt = 5123
> > thankyou = 5124
> > }
> >
> > 1
> > {
> > action = debitcardbalance
> > declines = debitcard-declines
> > pin-prompt = 5122
> > }
> >
> > 2
> > {
> > action = debitcardgenerate
> > declines = debitcard-declines
> > cardtype = 420
> > }
> >
> > 3
> > {
> > action = vmail
> > userno = 3641234
> > }
> >
> > Once it goes into vmail, we have a whole hard-coded voicemail app that
> > is basically modeled on the Centigram VoiceMemo system including
> > message sending, forwarding, paging, transferring to attendant.
> > Likewise for other functions. It also does queries into our backend
> > billing database and credit card charging systems.
> >
> > I wish I could convince my company to release the source code, but I
> > don't think that's ever going to happen, as they consider this code to
> > be their strategic advantage. It's been developed over many years and
> > used to run with Dialogic hardware but we have recently ported it to
> > asterisk.
> >
> > So yes, you can build almost any IVR application under asterisk. Not
> > only is the hardware one-eighth the cost, but the tech support is far
> > far superior. We used to wait years for Dialogic to fix bugs. And
> > best of all, asterisk provides all the core functions so you can spend
> > your time on your custom features and not on coding the basic
> > telephony functions, message playing, etc.
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
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>
___
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Re: [Asterisk-Users] Proposed IAX2 Name
or Packet Telephony (Simple) Protocol On 13 Mar 2003, Karl Putland wrote: > What about ITP > > Internet/IP > Telephony > Protocol > > On Thu, 2003-03-13 at 09:40, Mark Spencer wrote: > > > LIghtweight > > > Voice over IP > > > Exchange > > > > Or: > > > > Lightweight > > Internet > > Voice > > Exchange > > > > Mark > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Karl Putland <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to transfer a call??
Of courese: exten => 9998,1,Dial,SIP/9998|30|tTm Notice when you don't use the timeout you do have to use the options separator "|" like this: exten => 9998,1,Dial,SIP/9998||tTm but I think that T is not yet implemented regards Martin On Fri, 14 Mar 2003, WipeOut . wrote: > Thanks the 'show application dial' was useful.. > > Can multiple options be specified? > eg. exten => 9998,1,Dial,SIP/9998|30|t|T > > > > - Original Message - > From: Pertti Pikkarainen <[EMAIL PROTECTED]> > Date: Fri, 14 Mar 2003 15:15:14 +0200 > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] How to transfer a call?? > > > > > I have it like this > > > > exten => 9998,1,Dial,SIP/9998|30|t > > > > 30 is a timeout value > > Check 'show application dial' > > > > > > WipeOut wrote: > > > > >What is the correct syntax to use the 't' option?? > > > > > >This is the current line in my extensions.conf > > >exten => 9998,1,Dial,SIP/9998 > > >So would I change it to > > >exten => 9998,1,Dial,SIP/9998,t > > > > > >Thanks. > > > > > >- Original Message - > > >From: Pertti Pikkarainen <[EMAIL PROTECTED]> > > >Date: Fri, 14 Mar 2003 13:50:21 +0200 > > >To: [EMAIL PROTECTED] > > >Subject: Re: [Asterisk-Users] How to transfer a call?? > > > > > > > > > > > >>Negative side effect with 't' option: all the local SIP-to-SIP media > > >>streams travel trough Asterisk instead of going direct. > > >> > > >>Right now I'm using SNOM's transfer option instead. > > >>But now I can't use * call parking because of that. Using # is > > >>probably better > > >>if there are no scaling problems. > > >> > > >>Regards Pertti > > >> > > >> > > >> > > >>Steven Critchfield wrote: > > >> > > >> > > >> > > >>>If you search the archives you would find that for IP phone you need to > > >>>add a 't' option to the end of your dial command. The 't' option will > > >>>let the user dial '#' to get the systems attention, then dial an > > >>>extention for the transfer. > > >>> > > >>>On Fri, 2003-03-14 at 03:32, =?iso-8859-1?B?V2lwZU91dCCZ ?= wrote: > > >>> > > >>> > > >>> > > >>> > > >>>>Hi, > > >>>> > > >>>>Firstly let me start off by saying that asterisk is one of the most amazing > > >>>>pieces of open source I have seen, it rates right up there with Apache, > > >>>>OpenOffice, MySQL and even Linux itself.. Nice work!! > > >>>> > > >>>>I have just installed my first server, thanks to the astinstall script.. and I > > >>>>have read the Handbook (ver 1) and the white paper PDF's.. and I have managed > > >>>>to setup 2 extentions and make calls between them using MSN Messenger, nothing > > >>>>fantastic but its a start.. > > >>>> > > >>>>One answer is still missing.. How do I transfer a call to another ext?? I am > > >>>>looking at only using IP phones and so for the test system I am using MSN > > >>>>Messenger.. The final solution will probably use a linux softphone line > > >>>>gnophone or linphone.. > > >>>> > > >>>>All I have been able to find in the docs about call transfer is using a normal > > >>>>phone handset and hook-flash (not quite sure what that it, I am new to > > >>>>telephony).. > > >>>> > > >>>>So I guess what I am asking is what do I need to configure or do to be able to > > >>>>transfer a call from one IP ext to another?? > > >>>> > > >>>>Thanks.. > > >>>> > > >>>> > > >>>> > > >>>> > > >>___ > > >>Asterisk-Users mailing list > > >>[EMAIL PROTECTED] > > >>http://lists.digium.com/mailman/listinfo/asterisk-users > > >> > > >> > > > > > > > > > > > > > -- > > > > ** > > Nordic LAN&WAN Communication Oy > > Pertti Pikkarainen > > vp of engineering > > E-Mail: [EMAIL PROTECTED] > > tel: +358-9-5024100 > > fax: +358-9-5023840 > > gsm: +358-500-511467 > > > > Sinikalliontie 16 > > 02630 Espoo > > FINLAND > > > > ** > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > __ > http://www.linuxmail.org/ > Now with e-mail forwarding for only US$5.95/yr > > Powered by Outblaze > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec Formats
The formats that asterisk uses are #define'd in asterisk/include/asterisk/frame.h RTP formats are #define'd in asterisk/rtp.c regards Martin On Fri, 14 Mar 2003, John Vozza wrote: > I've been trying to find a list of codec "format numbers" so I can more > clearly understand the following message; > > Accepting AUTHENTICATED call from xxx.xxx.xxx.xxx, requested format = 4, > actual format = 4 > > I've seen 4, 32, 512 and I think a few others. For example I think format > 32 equal ADPCM but what are the others? > > TIA > > John > - > NetRom Internet Services 973-208-1339 voice > [EMAIL PROTECTED] 973-208-0942 fax > http://www.netrom.com > - > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No way to send secret...
Note that the message comes from chan_iax2.c that is under developement. It uses iax.conf as well as chan_iax.c regards Martin On Sat, 15 Mar 2003, John Vozza wrote: > Thanks to all who set me straight on the codec format stuff... > > I have a "remote" asterisk system running on my laptop which sucessfully > connects back to my main * server. (Lets me bring my phones to my customer > jobsites...) > > After the last few CVS updates I started seeing; > > NOTICE[13326]: File chan_iax2.c, Line 2999 (authenticate): No way to send > secret to peer 'xxx.xxx.xxx.xxx' (their methods: 4) > > Everything still works fine but should I be concerned about this or is my > iax.conf missing something? > > Thanks > > John > - > NetRom Internet Services 973-208-1339 voice > [EMAIL PROTECTED] 973-208-0942 fax > http://www.netrom.com > - > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ringdown Circuit Configuration
You have to add immediate=yes to zapata.conf to the declaration of this channel. Then right after someone picks up the phone asterisk will just right to 's' extension of the specified context. regards Martin On Tue, 18 Mar 2003, Don Pobanz wrote: > We have need of a ringdown circuit in an elevator. If someone picks up > the phone, it should dial another extension without any keys being > pressed. (There are no keys on the phone) > > If it was an incoming call to asterisk, the following lines in > extensions.conf would do the trick. > exten => s,1,Answer > exten => s,2,Dial,Zap/10 > > However, the 's' state is not valid for just picking up a phone > (extension). With nothing being dialed there are no extension matches > to make. It is like the dial tone needs to time out in a very short > time and instead of getting the busy tone, asterisk should dial an > extension. > > Anyway, I have not figured it out. Suggestions? > > Don Pobanz > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ${variable:a:b}
It's fixed now.
> what's the intended behaviour of ${variable:a:b}?
it's the same as substring application
> given that ${exten} = 501234
> until yesterday ${exten:2} would give '1234'
and it does now
>
> with current CVS ${exten:2} is '50' while ${exten:2:4} is '1234'
>
> how do I just strip characters/digits in front of the variable without
> specifying a length?
you can do ${variable:a} or ${variable:a:1000} :)
Martin
>
>
> I was doing things like:
> exten => _00.,1,Goto(context,${EXTEN:2},1)
>
> thanks,
> lele
>
>
>
>
> ___
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>
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Re: [Asterisk-Users] Use 1 port of an E400P as IP connection
Sure. You configure it as HDLC or CISCO HDLC and you have hdlc0 interface to send data. regards Martin On Thu, 20 Mar 2003, David Luyens wrote: > Hi, > > I would like to use * as a compression box. > Between 2 sites I have an E1 leased line. > > So would it be possible to use 1 port of an E400P card to route my IP > IAX calls over in order to cut the ip router out of the picture? > > David > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Use 1 port of an E400P as IP connection
Configure the span 1 as nethdlc=1-31 You have to have HDLC enabled in your kernel. Then compile zaptel/sethdlc (in zaptel "make sethdlc") and then ztcfg -vv sethdlc hdlc0 mode hdlc (or mode cisco) ifconfig hdlc ip_address network network_address that's it! Now you have IP over HDLC/CISCO HDLC. regards Martin On Thu, 20 Mar 2003, David Luyens wrote: > Thanks Martin, could you point me into the direction on how to do this? > > David > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko > Sent: Thursday, March 20, 2003 7:10 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Use 1 port of an E400P as IP connection > > > Sure. You configure it as HDLC or CISCO HDLC and you have > hdlc0 interface to send data. > > regards > Martin > > On Thu, 20 Mar 2003, David Luyens wrote: > > > Hi, > > > > I would like to use * as a compression box. > > Between 2 sites I have an E1 leased line. > > > > So would it be possible to use 1 port of an E400P card to route my IP > > IAX calls over in order to cut the ip router out of the picture? > > > > David > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound card and other warning messages
> Greetings Asterisk users. > > When I launch Asterisk, I get the following > > Asterisk CVS-03/20/03-16:56:24, Copyright (C) 1999-2001 Linux Support > Services, Inc. > Written by Mark Spencer <[EMAIL PROTECTED]> > = > [ Booting..-- Registered indication country 'us' > -- Registered indication country 'au' > -- Registered indication country 'fr' > -- Registered indication country 'de' > -- Registered indication country 'nl' > -- Registered indication country 'uk' > -- Setting default indication country to 'us' > ...WARNING[16384]: File chan_oss.c, Line 342 (setformat): Requested 8000 Hz, > got 8018 Hz -- sound may be choppy It means only that you have some cheap sound card. If it works for you that's fine. > WARNING[98311]: File chan_oss.c, Line 228 (sound_thread): Read error on > sound device: Resource temporarily unavailable If you don't want this messages put noload => chan_oss.so in /etc/asterisk/modules.conf > .WARNING[16384]: File chan_iax2.c, Line 4927 (set_config): Ignoring port for > now This channel driver is under developement. Also it reads the config files of chan_zap.c so you shouldn't bother about that. > /var/spool/asterisk/outgoing > .WARNING[16384]: File > chan_zap.c, Line 6416 (load_module): Ignoring rxwink It's connected with Atlas ... just comment out in zapata.conf > ] > Asterisk Ready. > *CLI> > > 1. The warning messages listed above, should I be concerned? > I tested the sound card, by entering the command "dial", it plays the > Asterisk greeting message on the speakers. I woudln't worry about yours. > > 2. Should I replace the sound card with a different one. Currently, it's a > ES1370. (Ensoniq AudioPCI) If it works for you => you don't have to. > > 3. After reading multiple docs, FAQ and mailing list archives, I am unable > to find out purpose of the sound card? Is it used to play music, while on > hold? Telephony PBX's usually don't have a sound card in them. You don't need sound card in your PBX. It may be used as another extension (read: another "phone") > > 4. The WARNING - chan_iax2.c, I think, this is a configuration problem > (IAX), from the default files. That I have not gotten to yet. THis channel driver does not use "port" keyword. regards Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reserving a minimum qty of channels
You could use a global variable like this:
[globals]
max_channels_1=10
max_channels_2=13
appl_1=0
[incoming1]
exten => _.,1,GotoIf,$[${appl_1} > ${max_channels_1}]?hangup|1:2
exten => _.,2,SetGlobalVar,appl_1=$[${appl_1} + 1]
exten => _.,3,do_what_you_want_to_do
exten => h,1,SetGlobalVar,appl_1=$[${appl_1} - 1]
exten => hangup,1,Congestion
exten => hangup,2,Hangup
and similar for [incoming_2]
regards
Martin
On Mon, 24 Mar 2003, Ivar van de Pieterman wrote:
> We have two vrs applications running on the same PRI, on different
> phonenumbers. All our traffic is incoming only.
>
> To make sure one application will not eat all the channels, I'd like to
> reserve a minimum of channels available per application. I don't want to
> give each app a fixed number of channels, bcz it could be a waste of
> resources.
>
> I've tried to find some more info about this in the current docs, but
> without success.
>
> Anyone ever tried this? Could you give me a direction where to look?
>
> Thanks, Ivar.
>
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Re: [Asterisk-Users] I100E - how far off is it??
I think it's all about SNOM 100 phone with IAX image. Martin On Mon, 24 Mar 2003, WipeOut . wrote: > Its still there.. > > http://www.asterisk.org/main/index.html > > Click "Hardware" on the left menu.. > > - Original Message - > From: Steve Kann <[EMAIL PROTECTED]> > Date: 24 Mar 2003 12:17:10 -0500 > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] I100E - how far off is it?? > > > > > Hmm, must have been a boo-boo, 'cause it seems to be gone already :) > > > > -SteveK > > > > On Mon, 2003-03-24 at 11:23, WipeOut . wrote: > > > I see on the new * website the following device listed under the > > > hardware section.. > > > > > > I100E: An IAX client Voice over IP telephone running linux. > > > > > > Anyone know anything more about it and when it will be available? > > -- > > Steve Kann - Chief Engineer - 520 8th Ave #2300 NY 10018 - (212) 533-1775 > > HorizonLive.com - collaborate . interact . learn > >"The box said 'Requires Windows 95, NT, or better,' so I installed Linux." > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > __ > http://www.linuxmail.org/ > Now with e-mail forwarding for only US$5.95/yr > > Powered by Outblaze > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] macros working?
It's there On Mon, 24 Mar 2003, Darrell Eldridge wrote: > Is the macro functionality (described in the draft > handbook Version 2, Section 4.3.11 Using Macros) > already available? I'm having trouble making it work > and wonder if I need to keep trying or wait until it's > in the code. > > Thanks, > > D. > > __ > Do you Yahoo!? > Yahoo! Platinum - Watch CBS' NCAA March Madness, live on your desktop! > http://platinum.yahoo.com > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Latest CVS causes compile time error
how about libpri ? On Tue, 25 Mar 2003, Lenny Tropiano / asterisk.org Mailing list wrote: > gcc -shared -Xlinker -x -o chan_phone.so chan_phone.o > gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations > -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 > -DASTERISK_VERSION=\"CVS-03/25/03-10:49:30\" -DINSTALL_PREFIX=\"\" > -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" > -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" > -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" > -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" > -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" > -Wno-missing-prototypes -Wno-missing-declarations -DIAX_TRUNKING -DCRYPTO -o > chan_zap.o chan_zap.c > chan_zap.c: In function `zt_digit': > chan_zap.c:677: warning: implicit declaration of function `pri_information' > chan_zap.c:677: structure has no member named `pri' > chan_zap.c:677: structure has no member named `call' > chan_zap.c: In function `zt_call': > chan_zap.c:1144: warning: unused variable `s' > make[1]: *** [chan_zap.o] Error 1 > > Yes, I have the latest "zaptel" too. Any ideas? > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe PIN functionality
You can try to do that using authenticate application regards Martin On Thu, 27 Mar 2003, James Golovich wrote: > > > > On 27 Mar 2003, Matthew Farley wrote: > > > My asterisk system is now working wonderfully (thanks to all of you for > > your invaluable contribution to the software world and your assistance > > on this list)! > > > > Enough of that... On to my current issue. MeetMe is working just fine in > > a basic sense, but when I try to assign a PIN to a conference room, it > > seems to make no difference in the result (the caller is placed in the > > conference without being asked to enter a PIN). > > > > > > > > > > My extension definition for the conference room looks like this: > > > > exten => 8600,1,Wait,1 > > exten => 8600,2,Playback(wstconfbeta) > > exten => 8600,3,Meetmecount,8600 > > exten => 8600,4,Meetme,8600|p|1234 > > > > My meetme.conf looks like: > > > > conf => 8600 > > > > > > > > > > Have I messed up somewhere in the config for this, or is there something > > not quite right with the PIN functionality of MeetMe? Any advice you can > > offer would be greatly appreciated. > > > > The meetme pin functionality is not implemented yet. As well as the > meetme admin menu. > > I don't think I'll have enough to implement this anytime soon, so maybe > someone else can finish the code up. > > James > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem Recording GSM file
You have to call record like this
exten => 8000,1,Record,intro:gsm
(read show application record)
Martin
On Thu, 27 Mar 2003, Michael K. Rodriguez wrote:
> This the error I receive when I try to record a GSM file
>
>
>
>
>
> -- Executing Record("SIP/67.98.37.220:5060", "intro|gsm") in new
> stack
>
> -- Playing 'beep'
>
> WARNING[15374]: File file.c, Line 602 (ast_writefile): No such format ''
>
> WARNING[15374]: File app_record.c, Line 143 (record_exec): Could not
> create file intro|gsm
>
>
>
>
>
> extension.conf
>
>
>
> exten => ,1,Record,intro|gsm
>
>
>
>
>
> Thanks
>
>
>
>
>
>
>
> _
>
>
>
> Michael K. Rodriguez
>
> DialMex LLC
>
> NOC Engineer
>
> 200 S. 10th Street Suite 1209
>
> McAllen, TX 78501
>
>
>
> (956) 994-0014 x107 office
>
> (956) 239-0627 mobile
>
> (956) 682-5821 fax
>
> [EMAIL PROTECTED]
>
>
>
> Escalation <http://www.dialmex.net/page> Procedure
>
> +++The information transmitted is intended only for the person or entity
> to which it is addressed and may contain confidential and/or privileged
> material. Any review, retransmission, dissemination or other use of, or
> taking of any action in reliance upon, this information by persons or
> entities other than the intended recipient is prohibited. If you
> received this in error, please contact the sender and destroy any copies
> of this document.+++
>
>
>
>
___
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Re: [Asterisk-Users] Asterisk Backtrace
Do also "frame 0" Martin On Thu, 27 Mar 2003, Eric Wieling wrote: > I'm getting occasional seg faults when a call ends. Here is the > backtrace. Calls are going to a SIP device. > > (gdb) bt > #0 0x08057bae in ast_queue_frame (chan=0x8125240, fin=0xbf5fea0c, lock=1) > at channel.c:344 > #1 0x08057dbf in ast_queue_hangup (chan=0x8125240, lock=1) at channel.c:380 > #2 0x401fa8d0 in handle_request (p=0x811ed50, req=0xbf5ff2bc, sin=0xbf5ff2ac) > at chan_sip.c:3592 > #3 0x401fae78 in sipsock_read (id=0x80cd418, fd=7, events=1, ignore=0x0) > at chan_sip.c:3669 > #4 0x080511b8 in ast_io_wait (ioc=0x80cd450, howlong=1000) at io.c:268 > #5 0x401fb248 in do_monitor (data=0x0) at chan_sip.c:3763 > #6 0x40026e8e in pthread_start_thread (arg=0xbf5ffc00) at manager.c:284 > (gdb) > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] very strange hangup.
You can use "debug channel " on asterisk console to find out which channel sends disconnect/hangup You have to trace all the channels on both of your systems. regards Martin On Thu, 27 Mar 2003, diana wrote: > Hello, > > I have a very complicated system which contains 2 *'s. > > PSTN --- CISCO --- H.323 --- 1 Asterisk --- IAX --- 2 Asterisk --- PSTN > > In the middle of conversation i get a Hangup. I get this logs when it > disconnect on the 2 Asterisk. > I must metion that the call is originated by Cisco, and i can't make any > calls to the Cisco. > > Rx-Frame Retry[N/A] -- Seqno: 12 Type: IAX Subclass: ACK > > Protocol Discriminator: Q.931 (8) len=9 > > Call Ref: len= 2 (reference 6/0x6) (Originator) > > Message type: DISCONNECT (69) > > Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 > Location: Private network serving the local user (1) > > Ext: 1 Cause: Normal Clearing (16), class = Normal > Event (1) ] > -- Hungup 'Zap/3-1' > == Spawn extension (flux1, 11011, 1) exited non-zero on > '[EMAIL PROTECTED]/2' > Tx-Frame Retry[000] -- Seqno: 13 Type: IAX Subclass: HANGUP > -- Hungup '[EMAIL PROTECTED]/2' > Rx-Frame Retry[N/A] -- Seqno: 13 Type: IAX Subclass: ACK > < Protocol Discriminator: Q.931 (8) len=5 > < Call Ref: len= 2 (reference 32774/0x8006) (Terminator) > < Message type: RELEASE (77) > > Protocol Discriminator: Q.931 (8) len=5 > > Call Ref: len= 2 (reference 6/0x6) (Originator) > > Message type: RELEASE COMPLETE (90) > > > I really don't know what is going on, and i will apreciate any idee. > > Thank you very much. > > Diana Cionoiu > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 port FXS card
> Will to ports on this card be able to act as FXO as well, or just as FXS? Maybe later. But there was some posting about "FXS to FXO converter" a few weeks before ??? > If the answer is yes, can we control which ports do which in any > combination? Why not ? > Finally, can this card coexist with the X100P FXO card in the > same PC and will Asterisk support them all at the same time? Why not ? Martin > > Thanks... > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * + Codecs + Hardphones??
The same as you go over the number of PRI channels ? regards Martin On Thu, 27 Mar 2003, James O. Sizemore III wrote: > Quick question what happens if you go over > your channel licenses? > > Mark Spencer wrote: > > >>So it looks like the best codec is the GSM codec as far and badwidth > >>vs voice quality, but I can't seem to find which hard phones support > >>the GSM codec or if * supports the G.729 codecs or others.. > >> > >>Which phones do the * user commumity find work the best?? and which > >>codecs do you use?? > >> > >> > > > >You can purchase G.729 from Digium at $10/channel. Contact Greg Vance > >(256-428-6262). The G.729 is currently considered "beta". > > > >Mark > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP module load error
You must be using some old code. Try to use code from CVS. Instructions are on www.digium.com regards Martin On Fri, 28 Mar 2003, [ISO-8859-7] ÓôáìÜôçò ÊåêÝò wrote: > Hello everybody. > > I have a test box with asterisk and till now I have successfuly made it > work with iax. > I'm trying to load the SIP module in order to test SIP with msn > messenger but without any success. > In my modules conf when I have : > > chan_sip.so=yes > > The asterisk service starts but it does not load any SIP module. When I > remove that line and I place a line after the chan_modem.so which is the > following : > > load chan_sip.so > > then I recieve the following lines in my /var/log/asterisk/messages : > > Mar 28 10:00:57 WARNING[1024]: File loader.c, Line 212 > (ast_load_resource): /usr/lib/asterisk/modules/chan_sip.so: undefined > symbol: ast_moh_stop > Mar 28 10:00:57 WARNING[1024]: File loader.c, Line 319 (load_modules): > Loading module chan_sip.so failed! > > What happens with that ? > Is there any solution for that ? > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] very strange hangup.
I'm afraid you have to do "debug channel" also on [EMAIL PROTECTED]/3 (since that info doesn't tell me much) The same on the other box. regards Martin On Fri, 28 Mar 2003, diana wrote: > << [ TYPE: Null Frame (4) SUBCLASS: N/A (5) ] [Zap/1-1] > -- Hungup 'Zap/1-1' > == Spawn extension (flux1, 11011, 1) exited non-zero on > '[EMAIL PROTECTED]/3' > -- Hungup '[EMAIL PROTECTED]/3' > > I get this error message in 2 Asterisk, when the channel is closed. > Any idee about this? > > Diana Cionoiu > > > > You can use "debug channel " on asterisk console > > to find out which channel sends disconnect/hangup > > You have to trace all the channels on both of your systems. > > > > regards > > Martin > > > > On Thu, 27 Mar 2003, diana wrote: > > > > > Hello, > > > > > > I have a very complicated system which contains 2 *'s. > > > > > > PSTN --- CISCO --- H.323 --- 1 Asterisk --- IAX --- 2 Asterisk --- PSTN > > > > > > In the middle of conversation i get a Hangup. I get this logs when it > > > disconnect on the 2 Asterisk. > > > I must metion that the call is originated by Cisco, and i can't make any > > > calls to the Cisco. > > > > > > Rx-Frame Retry[N/A] -- Seqno: 12 Type: IAX Subclass: ACK > > > > Protocol Discriminator: Q.931 (8) len=9 > > > > Call Ref: len= 2 (reference 6/0x6) (Originator) > > > > Message type: DISCONNECT (69) > > > > Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 > > > Location: Private network serving the local user (1) > > > > Ext: 1 Cause: Normal Clearing (16), class = Normal > > > Event (1) ] > > > -- Hungup 'Zap/3-1' > > > == Spawn extension (flux1, 11011, 1) exited non-zero on > > > '[EMAIL PROTECTED]/2' > > > Tx-Frame Retry[000] -- Seqno: 13 Type: IAX Subclass: HANGUP > > > -- Hungup '[EMAIL PROTECTED]/2' > > > Rx-Frame Retry[N/A] -- Seqno: 13 Type: IAX Subclass: ACK > > > < Protocol Discriminator: Q.931 (8) len=5 > > > < Call Ref: len= 2 (reference 32774/0x8006) (Terminator) > > > < Message type: RELEASE (77) > > > > Protocol Discriminator: Q.931 (8) len=5 > > > > Call Ref: len= 2 (reference 6/0x6) (Originator) > > > > Message type: RELEASE COMPLETE (90) > > > > > > > > > I really don't know what is going on, and i will apreciate any idee. > > > > > > Thank you very much. > > > > > > Diana Cionoiu > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using asterisk as secondary PBX ?
> I would like to known if these "T2" links are related to the E1 > stuff that everybody talk about on this forum. In other words, can I If you're in Europe than your T2 are 99.9% E1's (30 voice channels + 1 signalling) > link the free "T2" card of the Bosch to a Linux box with an E100P > interface and make it a secondary PBX with Asterisk ? One which will > be aware of VoIP ? Yes. You would have to modify the routing on Bosch to be aware of Asterisk. regards Martin > > Any hint will be very much appreciated ... > > Xavier > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] kphone registration failures
You shouldn't have problems in a recent CVS with that. regards Martin On Fri, 28 Mar 2003, Brian Capouch wrote: > Eric Wieling wrote:> Try commeting out the username= and secret= and set the > > host=dynamic. If that works you can try adding them back in. > > > > That did work!! > > But (sigh. . . ) on to the next problem. Asterisk doesn't like the SIP > messages that kphone is sending: > > WARNING[114696]: File chan_sip.c, Line 1431 (process_sdp): Error in > codec string 'a=rtpmap:0 PCMU/8000' > > I get the two interfaces bridged together, but I get silence going from > the kphone side to the asterisk side. . . > > Tweak asterisk, I wonder, or tweak kphone? > > Thanks. > > B. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - Hardware needed
What signalling are you going to use ? regards Martin On Mon, 31 Mar 2003, Eduardo Goncalves wrote: > Hi, > > I'm abaut to install asterisk and I want to know if buying an E400P (Quad Span E-1 > Interface) from digium my linux box will be ready (of course, after configure it) to > work with PSTN and my already in use PBX (E1 interface), or I'll need to buy > additional DSP's cards or whatever? > > > thanks > > Eduardo Gonçalves > AceNet do Brasil LTDA > 55 11 3365-0500 > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Config??
Read www.digium.com at Documentation->FAQ Martin On Mon, 31 Mar 2003, Allan Wang wrote: > Steven, > > > Could you please give me your config files for X100P > and S100U? I just got mine, but I have difficulty to > make it work. > Thank you. > > > > > Allan > > > > --- Steven Critchfield <[EMAIL PROTECTED]> wrote: > > On Fri, 2003-03-21 at 10:07, WipeOut . wrote: > > > I have got my box up and running with a X100P and > > a S100U > > > but I found a bit of a "funny".. > > > > > > I took the default config files and commented > > every line > > > and then I started creating my own config using > > the > > > commented out lines for reference.. (best way to > > learn) > > > > > > None of my configs worked and I could not work out > > why.. > > > > > > I tried everthing with out any luck.. > > > > > > So I cut my config lines from the bottom of the > > file to > > > the top, and restarted *.. Guess what?? It now > > worked > > > perfectly.. > > > > > > Is there a reason for this?? > > > > I'd say there was a problem in the commenting > > somewhere. I experienced a > > problem where I had a single charecter typo that > > made asterisk at that > > time segfault. I'm pretty sure that isn't the case > > anymore, but it may > > have caused the parser to stop parseing, and > > therefore not get to your > > config. Of course this is just a guess. > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > __ > Do you Yahoo!? > Yahoo! Platinum - Watch CBS' NCAA March Madness, live on your desktop! > http://platinum.yahoo.com > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_zap.c Warning : channel already in use
cvs update your libpri regards Martin On Mon, 31 Mar 2003, Alex Zarubin wrote: > Hi, > > There are several channels on the PRI span with the periodic warning: > > WARNING[9226]: File chan_zap.c, Line 5437 (pri_dchannel): Ring requested on > channel 21 already in use on span 1. Hanging up owner. > > 1. Any known reason for this message? > 2. Is there a way to reset the channels in question (without resetting the > whole span)? > > Thank you > Alex > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - Hardware needed
E&M winkstart, loopstart and groundstart signalling should work without problems. We don't support Qsig signalling yet. regards Martin On Tue, 1 Apr 2003, Eduardo Goncalves wrote: > I can work with digital E&M - winkstart, immediate, loopstart, groundstart and > ISDN with Qsig. Also with R2, but here in Brasil I prefer the first. > > regards > Eduardo > > > On Mon, 31 Mar 2003 15:17:45 -0600 (CST) > Martin Pycko <[EMAIL PROTECTED]> wrote: > > > What signalling are you going to use ? > > > > regards > > Martin > > > > On Mon, 31 Mar 2003, Eduardo Goncalves wrote: > > > > > Hi, > > > > > > I'm abaut to install asterisk and I want to know if buying an E400P (Quad Span > > > E-1 Interface) from digium my linux box will be ready (of course, after > > > configure it) to work with PSTN and my already in use PBX (E1 interface), or > > > I'll need to buy additional DSP's cards or whatever? > > > > > > > > > thanks > > > > > > Eduardo Gonçalves > > > AceNet do Brasil LTDA > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How could I get * from CVS if I am not on theLinux platform?
then download the sources and compile it ... On Wed, 2 Apr 2003, it wrote: > I installed the cygwin yesterday. But it seems that the cygwin does not have > the cvs command. > > $ cvs > bash: cvs: command not found > > > Regards > john > > > - Original Message - > From: "Michael Bielicki" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]>; "duncan" <[EMAIL PROTECTED]> > Sent: Friday, March 28, 2003 2:10 AM > Subject: Re: [Asterisk-Users] How could I get * from CVS if I am not on the > Linux platform? > > > > why not use cygwin and normal cvs ? > > www.cygwin.com > > > > just my 2c > > cheers > > > > Michael > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line is stuck off hook...
Yes, we can do if's (GotoIf), +,_,*,/,%, etc ... like this
exten => t,2,SetVar,looptest=$[${looptest} + 1]
regards
Martin
On Tue, 1 Apr 2003, John Harragin wrote:
> Hey do we have the ability to incriment a variable?
> exten => t,2,SetVar,looptest=$((looptest + 1))
> I was thinking of doing a library of simple arithmetic and bash-like
> expansions for asterisk... like Zap/{1&,2&,3} - but it may already have this
> functionality.
>
> John
>
> > exten => t,1,GotoIf(${looptest},Hangup,1,1)
> > exten => t,2,SetVar,looptest=1
> >
> > [Hangup]
> > exten => 1,1,Hangup
> >
> >
> > > That would fix the problem for the most part, but why isn't *
> > > releasing the line when the caller hangs up.
> >
> > First, what type of signalling are you using? loopstart/groundstart/CPD?
> > It is important to know loopstart may not signal when it is hungup real
> > well. If the zhone fails to detect it, then it is possible for asterisk
> > to never know it hungup
> >
> >
> > > -Original Message-
> > > From: Steven Critchfield [mailto:[EMAIL PROTECTED]
> > > Sent: Tuesday, April 01, 2003 6:57 PM
> > > To: [EMAIL PROTECTED]
> > > Subject: Re: [Asterisk-Users] Line is stuck off hook...
> > >
> > >
> > > On Tue, 2003-04-01 at 17:13, Gene Kochanowsky wrote:
> > > > Greetings,
> > > >
> > > > I am running Asterisk with a T100P and a Zhone channel bank for over a
> > > > month now. For the most part it works fine but from time to time
> > > > (about once a week) the system will not let go of a line and will play
> > > > the greeting over and over. Anyone calling gets a busy signal. If I
> > > > reset Asterisk everything works fine. Has anyone seen this problem
> > > > before and fixed it? If so what did you do?
> > > >
> > > > > show version
> > > > Asterisk CVS-02/28/03-22:17:11 built by [EMAIL PROTECTED] on a i686 running
> > > > Linux.
> > > >
> > > > If there is anything else you need to know, let me know.
> > >
> > > What line signalling are you using?
> > >
> > > I'm assuming you are reffering to a phone line coming from the PSTN, is
> > > it analog, or is it digital?
> > >
> > > Have you tried soft hangup on it?
> > >
> > > Have you tried to place a timout on it, so asterisk finally times out
> > > the call?
> > >
> > > Maybe you would like to place a loop counter on the number of times the
> > > call makes it through the timeout extension, and after some threshold,
> > > you issue a hangup on the call.
> > --
> > Steven Critchfield <[EMAIL PROTECTED]>
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus.
>
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Re: [Asterisk-Users] segmentation fault
asterisk -vvvcg (use g option to generate the coredump file) than gdb asterisk core.pid bt Also you might send a log of "pri intense debug span " regards Martin On Wed, 2 Apr 2003, Alex Zarubin wrote: > Configuration: > Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown > P4 2.5 GHz, 1 GB RAM > T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s). > Each call gets transferred (Dial) to the SIP platform and stays for 5 min. > > Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days. > Segmentation fault. > Case 2. Asterisk built out of CVS Apr. 1. Test was running for 12 hours. > Segmentation fault. > > No coredump found. In case 1 there was a significant memory growth: > Top at the startup: > 15986 root 9 0 6440 6436 2144 S 0.0 0.6 0:00 asterisk > 15987 root 8 0 6440 6436 2144 S 0.0 0.6 0:00 asterisk > Top in several hours: > 15986 root 9 0 9192 9188 2148 S 0.0 0.9 0:00 asterisk > 15987 root 9 0 9192 9188 2148 S 0.0 0.9 0:00 asterisk > Top after a day: > 27441 root 9 0 45980 44M 2156 S 0.0 4.5 0:00 asterisk > 27442 root 8 0 45980 44M 2156 S 0.0 4.5 0:16 asterisk > Actually, I saw it over 50. > > There were some warning messages on the way. For example: > > Apr 1 23:22:33 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): > PRI: > Read on 86 failed: Unknown error 500 > Apr 1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): > PRI: > !! Got reject for frame 102, retransmitting frame 102 now, updating n_r! > Apr 1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): > PRI: > !! Got reject for frame 103, but we have nothing -- resetting! > Apr 1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): > PRI: > !! Got reject for frame 29, retransmitting frame 29 now, updating n_r! > Apr 1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): > PRI: > !! Got reject for frame 30, but we have nothing -- resetting! > Apr 1 23:30:00 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): > PRI: > Read on 87 failed: Unknown error 500 > Apr 1 23:30:24 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): > PRI: > Read on 86 failed: Unknown error 500 > > Question: > What do I do to give you more info? Should I issue 'ulimit -c unlimited' to > get a coredump? > Are there any flags/modes to set? > > Thank you. > Alex Zarubin > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Transfer
cvs update -r 1.x channels/chan_sip.c make install where 'x' is from 1 to 30 version 1.30 is dated 2003-04-02 if not sure check "rcs2log -v |more" regards Martin On Tue, 1 Apr 2003, Russ Beaupre, P.E. wrote: > A while ago SIP transfer via the # key on a call to a cell phone via > iconnect was working. I updated to the current CVS tonight and now that > functionality is gone. Any ideas as to how to enable it again? > > Thanks in advance > > -russ > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple X100P cards
You configure them as usual zaptel.conf fxoks=1-n #(n - how many cards you have) Then you can just plug a single phone line to each of them and then in zttool which one will go into OK from RED state. regards Martin On Thu, 3 Apr 2003, Jim Archer wrote: > Hi All... > > If I have more than 1 X100P card, how do I configure them so I know which > one is which? > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3player problem
Can you use Playback instead ?
Playback doesn't use mpg123.
regards
Martin
On Thu, 3 Apr 2003, Tamas Levente wrote:
> Hey,
> I've installed 0.59r mpg123 on a redhat box. I set the extension up for the
> mp3player. I called and it was playing the file back,but it was full of drops. like
> sound - silence - sound continued. I thought let's try with the developer version of
> mpg123 cos other extensions like echo test was working fine. so I've installed the
> 0.59s. Since then my * dies with this message:
>
> DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP to 0
> -- Executing Answer("SIP/levisnom-7efd", "") in new stack
> -- Executing MP3Player("SIP/levisnom-7efd", "/asterisk/c.mp3") in new stack
> Killed
>
> Any suggestion, how to solve this problem? Latest cvs of course.
>
> THX
> __
> Levente Tamás
> ICQ#: 13692773
> Current ICQ status:
> + More ways to contact me
> __
>
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Re: [Asterisk-Users] MP3player problem
I woudln't write that if it wouldn't support mp3.
On Thu, 3 Apr 2003, Tamas Levente wrote:
> And does playback support mp3?
> - Original Message -
> From: "Martin Pycko" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, April 03, 2003 5:58 PM
> Subject: Re: [Asterisk-Users] MP3player problem
>
>
> > Can you use Playback instead ?
> > Playback doesn't use mpg123.
> >
> > regards
> > Martin
> >
> > On Thu, 3 Apr 2003, Tamas Levente wrote:
> >
> > > Hey,
> > > I've installed 0.59r mpg123 on a redhat box. I set the extension up for
> the mp3player. I called and it was playing the file back,but it was full of
> drops. like sound - silence - sound continued. I thought let's try with the
> developer version of mpg123 cos other extensions like echo test was working
> fine. so I've installed the 0.59s. Since then my * dies with this message:
> > >
> > > DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP
> to 0
> > > -- Executing Answer("SIP/levisnom-7efd", "") in new stack
> > > -- Executing MP3Player("SIP/levisnom-7efd", "/asterisk/c.mp3") in
> new stack
> > > Killed
> > >
> > > Any suggestion, how to solve this problem? Latest cvs of course.
> > >
> > > THX
> > > __
> > > Levente Tamás
> > > ICQ#: 13692773
> > > Current ICQ status:
> > > + More ways to contact me
> > > __
> > >
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> ___
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Re: [Asterisk-Users] MP3player problem
codecs/codec_mp3_d.c -> xing ...
On Thu, 3 Apr 2003, Michael Bielicki wrote:
> what does playback use ?
>
> On Thursday 03 Apr 2003 16:18, Martin Pycko shaped the electrons to say:
> > Can you use Playback instead ?
> > Playback doesn't use mpg123.
> >
> > regards
> > Martin
> >
> > On Thu, 3 Apr 2003, Tamas Levente wrote:
> > > Hey,
> > > I've installed 0.59r mpg123 on a redhat box. I set the extension up for
> > > the mp3player. I called and it was playing the file back,but it was full
> > > of drops. like sound - silence - sound continued. I thought let's try
> > > with the developer version of mpg123 cos other extensions like echo test
> > > was working fine. so I've installed the 0.59s. Since then my * dies with
> > > this message:
> > >
> > > DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP
> > > to 0 -- Executing Answer("SIP/levisnom-7efd", "") in new stack
> > > -- Executing MP3Player("SIP/levisnom-7efd", "/asterisk/c.mp3") in new
> > > stack Killed
> > >
> > > Any suggestion, how to solve this problem? Latest cvs of course.
> > >
> > > THX
> > > __
> > > Levente Tamás
> > > ICQ#: 13692773
> > > Current ICQ status:
> > > + More ways to contact me
> > > __
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Michael Bielicki
> Managing Director
> TAAN Consultants Ltd
> http://www.global-gateway.net/
>
> --
>
> This correspondence is for the named person's use only. It may contain
> confidential or legally privileged information or both. No confidentiality
> or privilege is waived or lost by any mistransmission. If you receive this
> correspondence in error, please immediately delete it from your system and
> notify the sender. You must not disclose, copy or rely on any part of this
> correspondence if you are not the intended recipient.
>
> Any opinions expressed in this message are those of the individual sender.
>
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Re: [Asterisk-Users] FAX over IAX
Some people run fax over IAX using ulaw codec on the local LAN. Martin On Thu, 3 Apr 2003, Brian J. Schrock wrote: > From what I have heard packetizing fax does not work well, does not > matter if it is IAX or SIP. I think that was straight from digium tech > support. > > On Wednesday, April 2, 2003, at 09:53 AM, John Harragin wrote: > > > Hi, > > > > We are looking at consolidating our lines with PRI. This will allow the > > elimination of many fax lines. Some of them will be replaced with this > > type > > of config ... > > PRI * IAX * Channel-Bank FAX > > We will have daggressor suppressor enabled. Is anyone doing this and > > should > > I expect smooth operation? > > > > John > > > > > > This e-mail was scanned and found clean by Monroe-Woodbury CSD > > Antivirus. > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > Brian J. Schrock > Network Engineer, RHCE, CCNA > Anistone Technologies > Phone: 614-798-9106 > FAX: 614-573-7165 > 6926 Avery Rd. > Dublin, OH 43017 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] non-telephony use of T400P?
You could configure the channels of port 1 as clear=1-24 in zaptel.conf This way you'll have one big pipe on /dev/zap/1 accessible for you. Unfortunatelly you cannot do clear=1-96 and have it all 4 spans on /dev/zap/1. regards Martin On Thu, 3 Apr 2003, Chris Albertson wrote: > > Is there a block diagram or similar for the T400P? > What about programming information? What documentation > exists for it that a device driver writter might want? > > I need a T1 interface for a non-telephony application. > All I need to do is get the raw bits into a computer > > The application is this: > > A radio telemetry system sends a bit stream to a reciever > the reciever recovers the clock but not word boundries > it then puts this uninterpeted bit stream on one end of > a T1. ...(lots of detail omitted here.)... > At the other end we use "a bunch of equipment" to > get this data into a format usable by computers. I'd like > to replace this "bunch of equipment" with a PC, some > kind of a T1 interface card and some software. The more > basic and programable the T1 interface card the better as > the data is unconventional (from the viewpoint of a > PC) and might contain 12, 6, or 10 bit bytes mixed in > the same bit stream. Sometimes the bit stream is multiplexed > over multiple T1s. So I do _not_ need the T1 interface > card to do any protocol, just pass the bits. > > > = > Chris Albertson > Home: 310-376-1029 [EMAIL PROTECTED] > Cell: 310-990-7550 > Office: 310-336-5189 [EMAIL PROTECTED] > KG6OMK > > __ > Do you Yahoo!? > Yahoo! Tax Center - File online, calculators, forms, and more > http://tax.yahoo.com > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Valiant Comms VCL 30 Channel bank + DigiumE100P
Do you really have the channels in asterisk ? "zap show channels" Is the alarm on the E1 circuit ? Martin On Thu, 5 Jun 2003, Jay Banda wrote: > Hello All. > > Does anyone have experience with the Valiant Comms vcl30 channel > and the Digium E100P in asterisk ? We have the vcl30 channel bank, > loaded with FXO interfaces. We have set up * for fxs in zaptel.conf > and in zapata.conf, but are not able to get any incoming calls. > > The vcl fxs interfaces show that they are ringing ( incoming call from > PSTN ) , but the * does not answer ( or am I missing something , I think > the E100P card is supposed to indicate that there is an incoming call ?) > > If anyone is able to point me in a remotely correct direction, I will be > eternally grateful :) > > Kind regards to all > > Jay Banda > Snr Network Eng > CopperNET Solutions > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] email notification not working anymore
It does use sendmail. Which app are you using ? voicemail or voicemail2 ? Martin On Thu, 5 Jun 2003, Derek Beaumont wrote: > I used to have email notification working with my voicemail services but > it stopped working when I installed the new version of asterisk. > > I have not changed my voicemail.conf file, so I'm out of ideas. > > Does asterisk use Sendmail to send messages, or does it have its own > method for sending email? > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] answering calls with SIP phones
Yes, it is. Sip supports callpickup's with *8 same as zaptel hardware. You just need to configure it in sip.conf. regards Martin On Thu, 5 Jun 2003, John Todd wrote: > > I haven't tried this SIP features, but in the latest sip.conf.sample, > this is included: > > ;[pingtel] > ;type=friend > ;username=pingtel > ;secret=blah > ;host=dynamic > ;qualify=1000 ; Consider it down if it's 1 second to reply > ;callgroup=1,3-4 > ;pickupgroup=1,3-4 > ;defaultip=192.168.0.60 > > > Looks like that is somehow indicating that SIP supports call groups. > Is this incorrect? > > JT > > > > >Thanks, very good insights. > > > >The proposed method has a single flaw - it's very difficult to detect > >that all SIP channels are busy, and thus queue the call. > > > >It's a petty that SIP does not support call groups, it would make it > >automatic. > > > >Best, > > > >PHM > > > > > >-Original Message- > >From: [EMAIL PROTECTED] > >[mailto:[EMAIL PROTECTED] On Behalf Of > >[EMAIL PROTECTED] > >Sent: June 05, 2003 2:25 PM > >To: [EMAIL PROTECTED] > >Subject: Re: [Asterisk-Users] answering calls with SIP phones > > > >On Thu, 5 Jun 2003, Paulo Mannheimer wrote: > > > >> I have an incoming call that I would like answered every time by a > >> different SIP phone (out of 50). > > > >hmm... pass the call through an AGI first, that picks a random number > >and then pass the call to that SIP phone number > > > >> Also, some of the phone may not be available (may be turned off and > >thus > >> unregistered with Asterisk). > > > >set the unavailable priority to go back to the AGI for another try > > > >> Any way of doing this? > > > >i'm sure there are more, and intelligent ways of doing this > >but if you want a kludge, this'll work > > > >- wasim > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Parking on 7960
It should be exten=>sip,1,Dial(SIP/sipphone)||t Martin On 6 Jun 2003, Dave Wolven wrote: > Hi Don't know if someone answered this yet... > > when calling the dialapp append the |t to it > > exten=>sip,1,Dial(SIP/sipphone)|t > > This will allow you to hit # and then the callparking extension. > > Thanks > Dave > > On Tue, 2003-06-03 at 09:10, denon wrote: > > Hi all, > > > > I've got a fairly minor question, but it's getting on my nerves .. > > hopefully it's an easy answer. I'm having trouble parking calls on our > > 7960s. It works fine on ZAP devices, though, and they're both using the > > same context. > > > > What I do is: > > When I'm on the call, I hit More, Transfer, 700, Dial. I then get a fast > > busy. If I dial 701, for example, I do get the "no call parked here", so I > > know it's somewhat working. If I try a blind transfer, it just says it failed. > > > > Do I need to add an explicit extension with a ParkAndAnnounce for these? If > > so, anyone have a sample of what they use? > > > > Thanks for any direction, > > > > -d > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] more about SIP ...
You need to have
disallow=all
allow=g723.1
and the other remote phone has to use also the G723 codec. Otherwise
asterisk will try to transcode but it doesn't have the G.723 code itself.
regards
Martin
On Fri, 6 Jun 2003, Dave Alan Caruana wrote:
> I added the line "allow G723.1" in my sip.conf general config,
> and from a bridge connection which gives silence,
> I have progressed to the error message below,
> and the call gets rejected.
>
> help!!
>
> Dave
>
> ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant
> Expressa
> [EMAIL PROTECTED] : Go2Call SIP gateway
>
>
>
> -- Executing Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]")
> in new stack
> -- Called [EMAIL PROTECTED]
> WARNING[1240577216]: File channel.c, Line 1711
> (ast_channel_make_compatible): No path to translate from
> SIP/216.52.153.207-2e12(1) to SIP/217.168.168.49:5060(4)
> -- SIP/216.52.153.207-2e12 answered SIP/217.168.168.49:5060
> WARNING[1240577216]: File channel.c, Line 1711
> (ast_channel_make_compatible): No path to translate from
> SIP/217.168.168.49:5060(4) to SIP/216.52.153.207-2e12(1)
> WARNING[1240577216]: File app_dial.c, Line 606 (dial_exec): Had to drop call
> because I couldn't make SIP/217.168.168.49:5060 compatible with
> SIP/216.52.153.207-2e12
> == Spawn extension (default, 1303, 1) exited non-zero on
> 'SIP/217.168.168.49:5
>
>
>
>
>
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Re: [Asterisk-Users] Problem w/ Zaptel HDLC mode cisco Data Stability
What network card are you using ? (model and vendor) Martin On Tue, 27 May 2003, Nick Eggleston wrote: > We are using the zaptel driver to deliver a combined voice/data T1 circuit. > > The data channel-group is using the "cisco" hdlc protocol (on the linux side) > and connects with a cisco router on the far side. > > Everything comes up and works fine for a while. A while later, though, IP > communication fails completely. The cisco side shows up/up. The Linux side > has a message in the log file stating: "hdlc0: Link down". > > Just prior to the link down message are numerous messages that say "New offset:" > followed by different numbers. That may or may not be relevent. > > I am curious to know other people results with this configuration, if any. If > you had the same problem, how did you solve it? > > We are planning to switch to PPP and see if that improves the situation. > > --Nick > > (implementation details follow:) > > CISCO: > > interface Serial1/0:1 > ip address 1.0.0.102 255.255.255.252 > encapsulation hdlc > end > > LINUX: > > sethdlc hdlc0 mode cisco > ifconfig hdlc0 1.0.0.101 netmask 255.255.255.252 > > OS: Redhat 8.0 > Kernel: 2.4.18-27.8.0 > > /etc/zaptel.conf: > > loadzone=us > defaultzone=us > span=1,1,0,esf,b8zs > nethdlc=1-6 > bchan=13-23 > dchan=24 > > ZAPTEL driver CVS diff: > > cvs server: Diffing . > Index: Makefile > === > RCS file: /usr/cvsroot/zaptel/Makefile,v > retrieving revision 1.5 > diff -r1.5 Makefile > 36c36 > < #KFLAGS+=-DCONFIG_ZAPTEL_MMX > --- > > KFLAGS+=-DCONFIG_ZAPTEL_MMX > 54c54 > < #KFLAGS+=-DCONFIG_ZAPATA_NET > --- > > KFLAGS+=-DCONFIG_ZAPATA_NET > > ZAPTEL driver CVS/Entries: > > /.cvsignore/1.1.1.1/Wed Feb 12 13:59:20 2003// > /ChangeLog/1.1.1.1/Wed Feb 12 13:59:20 2003// > /README.fxsusb/1.1.1.1/Wed Feb 12 13:59:20 2003// > /arith.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /biquad.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /bittest.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /complex.cc/1.1.1.1/Mon Mar 17 18:11:45 2003// > /complex.h/1.1.1.1/Mon Mar 17 18:11:45 2003// > /digits.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /ecdis.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /fasthdlc.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /fir.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /fxstest.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /genconst.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /gendigits.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /hdlcgen.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /hdlcstress.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /hdlctest.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /hdlcverify.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /ifcfg-hdlc0/1.1.1.1/Wed Feb 12 13:59:20 2003// > /ifup-hdlc/1.1.1.1/Wed Feb 12 13:59:20 2003// > /makefw.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /mec.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /mec2.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /mec2_const.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /mec3.h/1.3/Wed Apr 16 03:31:07 2003// > /mkfilter.h/1.1.1.1/Mon Mar 17 18:11:45 2003// > /mknotch.cc/1.1.1.1/Mon Mar 17 18:11:45 2003// > /orig.ee/1.1.1.1/Wed Feb 12 13:59:20 2003// > /patgen.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /patlooptest.c/1.1.1.2/Sat Mar 15 06:00:30 2003// > /pattest.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /proslic.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /sec-2.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /sec.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /sethdlc.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /timertest.c/1.1/Thu Mar 20 07:00:04 2003// > /tonezone.c/1.2/Mon Apr 14 16:14:55 2003// > /tonezone.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /tor.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /tor2-hw.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /tor2.c/1.1.1.5/Mon Mar 17 18:11:45 2003// > /tor2.ee/1.1.1.1/Wed Feb 12 13:59:20 2003// > /tor2ee.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /torisa.c/1.1.1.2/Mon Mar 17 18:11:45 2003// > /torisa.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /torisatool.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /tormenta2.rbt/1.1.1.1/Wed Feb 12 13:59:20 2003// > /tormenta2.ucf/1.1.1.1/Wed Feb 12 13:59:20 2003// > /tormenta2.vhd/1.1.1.1/Wed Feb 12 13:59:20 2003// > /usbfxstest.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /wcfxo.c/1.5/Thu Apr 10 21:02:54 2003// > /wcfxs.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /wcfxsusb.c/1.1.1.1/Wed Feb 12 13:59:20 2003// > /wcfxsusb.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /wct1xxp.c/1.1.1.4/Mon Mar 17 18:11:45 2003// > /wcusb.c/1.1.1.2/Fri Mar 14 06:00:34 2003// > /wcusb.h/1.1.1.1/Wed Feb 12 13:59:20 2003// > /zaptel.conf.sample/1.1.1.2/Mon Mar 17 18:11:45 2003// > /zaptel.init/1.1.
Re: [Asterisk-Users] Bridging two iconnect calls
1) you need two accounts in iconnecthere 2) you need to register with two accounts 3) then simply receive the call using one and send it over another account Martin On Wed, 28 May 2003, pradeep kumar wrote: > Hi All. > > I am trying to setup asterisk so that I can place two outbound calls via > iconnecthere and connect them. > Is this possible ? If this is the case, please let me know what I need in > the extension conf to accomodate this feature. > > Thanks in advance > > PS : I did send a similar mail yesterday, but it does not seem to have > reached the mailing list. If this appears to be a duplicate please ignore. > > Pradeep > > _ > Reconnect with old pals. Relive the happy times. > http://www.batchmates.com/msn.asp With just one click. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
