[Asterisk-Users] oh323 on @homeasterisk
Anybody here added oh323 to @homeasterisk? I have compiled and add the oh323. I am wondering if anybody able to add the oh323 under web interface AMP? If anybody did it or know how to do it, please let me know. It has option for sip, IAX.. why not add h323 !! Thanks ** C.M. Rahman Jr. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Memory Requirements
On Fri, Apr 08, 2005 at 07:01:08PM -0500, Eric Rees wrote: I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB of memory. This is serving about 75 sip clients, Polycom500's and 600's. We are running into problems with the memory. Asterisk, right now, is using about 1.8GB of system memory. I am using Asterisk 1.0.7, Zaptel 1.0.7 with Digiums TE410 1xT1 RBS and 1xT1 PRI, Libpri 1.0.7 on Fedora Core 3. My question; is this normal or do I need more memory or is there a more serious underlying problem. How are measuring Asterisk memory usage? You're not counting the memory consumed by the filesystem cache, are you? Cam ___ 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] g726 gsm not working with sipura
I have a DID from livevoip coming into * as SIP/gsm.My phone is a sipura 2000 and the sip.conf for the sipura only allows g726. When I dial the sipura on an incoming call to connect the channels the sipura returns the error Media Type Not Available. If I set sip.conf to allow ulaw for the sipura it works fine. Following is the sip debug. What am I missing here? Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.160.244.186:5060;branch=z9hG4bK36dd7751;rport From: 4258201020 sip:[EMAIL PROTECTED];tag=as0163cfd3 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Live VoIP CJ.M-250 Date: Sat, 09 Apr 2005 07:01:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 345 v=0 o=root 18516 18516 IN IP4 217.160.244.186 s=session c=IN IP4 217.160.244.186 t=0 0 m=audio 13448 RTP/AVP 0 4 3 8 18 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 12 headers, 15 lines Using latest request as basis request Sending to 217.160.244.186 : 5060 (non-NAT) Found no matching peer or user for '217.160.244.186:5060' Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 217.160.244.186:13448 Found description format PCMU Found description format G723 Found description format GSM Found description format PCMA Found description format G729 Found description format iLBC Found description format telephone-event Capabilities: us - 0x2 (gsm), peer - audio=0x50f (g723|gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 8002703805 in default list_route: hop: sip:[EMAIL PROTECTED] Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.160.244.186:5060;branch=z9hG4bK36dd7751 From: 4258201020 sip:[EMAIL PROTECTED];tag=as0163cfd3 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 217.160.244.186:5060 -- Executing Wait(SIP/217.160.244.186-08688000, 3) in new stack -- Executing Answer(SIP/217.160.244.186-08688000, ) in new stack We're at 69.25.136.31 port 12976 Answering with preferred capability 0x2 (gsm) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 217.160.244.186:5060;branch=z9hG4bK36dd7751 From: 4258201020 sip:[EMAIL PROTECTED];tag=as0163cfd3 To: sip:[EMAIL PROTECTED];tag=as0c5f43fb Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 215 v=0 o=root 27417 27417 IN IP4 69.25.136.31 s=session c=IN IP4 69.25.136.31 t=0 0 m=audio 12976 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 217.160.244.186:5060 -- Executing Ringing(SIP/217.160.244.186-08688000, ) in new stack -- Executing Dial(SIP/217.160.244.186-08688000, SIP/chris|10) in new stack We're at 69.25.136.31 port 14790 Answering with capability 0x10 (g726) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 69.25.136.31:5060;branch=z9hG4bK77aa2d9a;rport From: 4258201020 sip:[EMAIL PROTECTED];tag=as179b5cc8 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sat, 09 Apr 2005 07:02:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 223 v=0 o=root 27417 27417 IN IP4 69.25.136.31 s=session c=IN IP4 69.25.136.31 t=0 0 m=audio 14790 RTP/AVP 111 101 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 4.32.7.39:50350 -- Called chris catalog1*CLI Sip read: SIP/2.0 100 Trying To: sip:[EMAIL PROTECTED]:5060 From: 4258201020 sip:[EMAIL PROTECTED];tag=as179b5cc8 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 69.25.136.31:5060;branch=z9hG4bK77aa2d9a Server: Sipura/SPA2000-2.0.13(g) Content-Length: 0 8 headers, 0 lines catalog1*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.160.244.186:5060;branch=z9hG4bK30b231db;rport From: 4258201020 sip:[EMAIL PROTECTED];tag=as0163cfd3 To: sip:[EMAIL PROTECTED];tag=as0c5f43fb Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Live VoIP CJ.M-250 Content-Length: 0 9 headers, 0 lines catalog1*CLI
Re: [Asterisk-Users] iax / realtime problems
Hi Mat, I can easily replicate the problem. I just put an entry on the iax table for mysql, fire up iax soft client and BOOM .. asterisk core dumps. What's weird is sip is working fine using realtime. Here is a gdb backtrace. Not really a programmer. Hope someone helps. Thanks. #0 0x00beeec0 in vfprintf () from /lib/tls/libc.so.6 (gdb) bt #0 0x00beeec0 in vfprintf () from /lib/tls/libc.so.6 #1 0x00c0f286 in vsnprintf () from /lib/tls/libc.so.6 #2 0x00bf7622 in snprintf () from /lib/tls/libc.so.6 #3 0x0031187a in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #4 0x00b19340 in ?? () #5 0x0100 in ?? () #6 0x00330900 in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #7 0x00b19480 in ?? () #8 0x0082d5d6 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #9 0x003308fc in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #10 0x00b19720 in ?? () #11 0x0031174c in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #12 0x in ?? () On Apr 8, 2005 8:53 PM, Matt Schulte [EMAIL PROTECTED] wrote: I've never actually core dumped but I *have* been able to hang asterisk a couple times, I believed my problem was when I lost my mysql connection. Why it lost connection is a mystery, the servers are on the same testswitch. :/ I forgot which head ver it was, a couple weeks ago. -Original Message- From: Paul P. Pongco [mailto:[EMAIL PROTECTED] Sent: Friday, April 08, 2005 1:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] iax / realtime problems Hello, I am using CVS-NHEAD-03/29/05-15:51:16 and testing iax realtime. I have configured a test account on iax.conf: [test] type=friend context=test username=test auth=md5 secret=testing host=dynamic disallow=all allow=ilbc allow=gsm callerid=1010 trunk=no qualify=no Then I insert an entry on mysql for testing realtime (btw realtime on the asterisk box works well for sip on both the flatfile and mysql). It has the same config as that on the flatfile but with different username and password (iaxtest). Asterisk crashes with the following error: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 03403 DCall: 0 [x.x.0.93:4569] USERNAME: iaxtest REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 3 DCall: 03403 [x.x.0.93:4569] -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 --snip, above lines just repeat here-- -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) On iax.conf rtcachefriends=yes rtnoupdate=yes rtautoclear=yes What could be causing this? Anyone seen this problem before? Help would be appreciated. Thanks. -- Cheers, Paul P. Pongco Mosaic Communications Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g726 gsm not working with sipura
snacktime wrote: I have a DID from livevoip coming into * as SIP/gsm.My phone is a sipura 2000 and the sip.conf for the sipura only allows g726. When I dial the sipura on an incoming call to connect the channels the sipura returns the error Media Type Not Available. If I set sip.conf to allow ulaw for the sipura it works fine. Following is the sip debug. What am I missing here? The fix for this was posted on the mailing list on Thursday. Look for the #define for the DEPRECATED g726 codec ID. Enable that define and recompile. B. ___ 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] g726 gsm not working with sipura
On Apr 9, 2005 12:14 AM, Brian Capouch [EMAIL PROTECTED] wrote: snacktime wrote: I have a DID from livevoip coming into * as SIP/gsm.My phone is a sipura 2000 and the sip.conf for the sipura only allows g726. When I dial the sipura on an incoming call to connect the channels the sipura returns the error Media Type Not Available. If I set sip.conf to allow ulaw for the sipura it works fine. Following is the sip debug. What am I missing here? The fix for this was posted on the mailing list on Thursday. Look for the #define for the DEPRECATED g726 codec ID. Enable that define and recompile. Yep that was it. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astcc Patch
I have installed Astcc and everything works fine. Except one issue, right after the card number is entered, Astcc prematurly sets the 'inuse' field in the 'cards' table to 1 to indicate someone is using the card. So if i entered the card number and i then hangup without dialing any number, Astcc has its inuse=1 permenantly and i will never be able to make a connection using that card number on any subsequent attempt. i could delete the inuse field, but then more than one person could use the same card and Astcc is not designed to handle multiple calls on the same card number. Pleas help. __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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] UK ISDN with Asterisk
Hi Henry staff member can take the call). If there is another way to do this with anaologue lines, i'm open to suggestions. I have looked at using a You used to be able (still can probably) do this with a thing called auxiliary working from BT on analogue lines. Two lines with one number, the second would take the call if the first were busy. I'd be tempted to steer clear of analogue though unless you really want it - I'm not sure how well it works directly hooked into the Asterisk server, but analogue on the voice gateways I've configured is just hard work a lot of the time. Good luck Dunc ___ 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: Lag in meetme
In article [EMAIL PROTECTED], BJ Weschke [EMAIL PROTECTED] wrote: What version of * are you running? There was a bug that was posted a few weeks back where when not using the q option it was possible for legs of the conference to get further separated from each other (sometimes up to 3-5 secs after 60 mins of conferencing). The bug is in mantis and there is a patch that seems to keep that down to no more than 1/2 to 3/4 second after several hours of conferencing, but I'm not certain that this patch made it back into the CVS-HEAD branch. It didn't (I was the proponent of the patch). I disagreed with Mark's alternative solution, and still do, so it's sort of stalled for now. The patch plays tones and announcements destined for the whole conf using a separate thread. Playing and waiting for them inline causes the channel's own queue not to be serviced, developing a backlog. This applies mainly to enter/leave sounds and 'Fred has joined' announcements. I believe there are other lag issues in non-Zap meetme that are not addressed by my patch, but I haven't had time to investigate them thoroughly yet. If I set up a meetme between two SIP phones with the 'q' flag set, I start off with negligible delay. If I just leave the conference running, that delay builds, up to the 1/2 to 3/4 of a second as reported above, but then sometimes disappears before building again. I haven't yet determined whether the lag is on the SIP-conf leg or the conf-SIP leg, or both. My suspicion is that it is drift due to clock differences between the ZAP timer and the SIP device, but am not sure how to determine when a frame should be dropped or duplicated. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and RT (Request Tracker) setup?
Thanks for the replies. I've got a bit further now after running the agi script manually, turns out the asterisk perl module was missing. I've got that and it gets somewhat further now. However I am running RT 3.4.1, so I assume by what you say Kris its probably not going to work? As soon as I leave a message now on the * console I see:- -- Executing Answer(SIP/2010-8f2c, ) in new stack -- Executing AGI(SIP/2010-8f2c, /var/lib/asterisk/agi-bin/rt_ticket.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/rt_ticket.agi -- AGI Script Executing Application: (Answer) Options: () -- Playing 'rt/ticket-choices' (language 'en') -- AGI Script Executing Application: (Record) Options: (/tmp/filesflYBL:wav) -- Playing 'beep' (language 'en') rt*CLI [Sat Apr 9 08:50:13 2005] [crit]: Can't call method IsPassword on an undefined value at /tmp/0.1/rt-soap-server.pl line 237. (/usr/local/rt3/lib//RT.pm:285) -- AGI Script /var/lib/asterisk/agi-bin/rt_ticket.agi completed, returning 0 And if I try to update an exisiting ticket I see: -- Executing Answer(SIP/2010-da15, ) in new stack -- Executing AGI(SIP/2010-da15, /var/lib/asterisk/agi-bin/rt_ticket.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/rt_ticket.agi -- AGI Script Executing Application: (Answer) Options: () -- Playing 'rt/ticket-choices' (language 'en') -- Playing 'rt/enter-ticket' (language 'en') -- AGI Script /var/lib/asterisk/agi-bin/rt_ticket.agi completed, returning 0 It's a shame it does not work, as I see a some nice possibilites with * and RT and maybe festival for reading some stuff back. Cheers Mike On Apr 9, 2005 4:41 AM, Kris Boutilier [EMAIL PROTECTED] wrote: Your success may depend on your version of RT. The RT-Soap-Server.pl interface has been depreciated. We've taken the somewhat easier, if less elegant approach of just setting up a voicemail box with a longwinded greeting that invites the caller to leave a VM that is in turn emailed as an attachment into RT: /etc/asterisk/extensions.conf: 6808 = ,Request Tracker,[EMAIL PROTECTED],,attach=yes|delete=yes ; Voicemails left in this box are forwarded into RT Certainly an interactive IVR would be much nicer, but it'll probably take a fair amount of work to construct against RT 3.4.x. For the uninitiated, Request Tracker is an elegant and light weight GPL'ed trouble ticketing system: http://www.bestpractical.com Great for managing all those user interactions stemming from a new deployment of Asterisk. :-) Kris Boutilier Information Systems Coordinatior Sunshine Coast Regional District -Original Message- From: Mike Dent [mailto:[EMAIL PROTECTED] Sent: Friday, April 08, 2005 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and RT (Request Tracker) setup? Hi, I'm attempting to get Asterisk and RT working, I've followed the details from http://megaglobal.net/docs/asterisk/html/rtasterisk.html When I dial the extension I put the rt agi on for testing all I get is silence and on the * console I see: rt*CLI -- Executing Answer(SIP/2010-f6f5, ) in new stack -- Executing AGI(SIP/2010-f6f5, /var/lib/asterisk/agi-bin/rt_ticket.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/rt_ticket.agi -- AGI Script /var/lib/asterisk/agi-bin/rt_ticket.agi completed, returning 0 Anybody else got this working? Thanks Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell suggestions for Quad T1 system
When I do see problems, It happens near capacity. All of the systems I will be running will be using all 92 lines when they are being used. Has anyone tested any of the new Dell systems in a production environment with this kind of load? Specifically I am looking at the 1850 and 2850. We have multiple Dual Xeon 3.6 Poweredge 1850s with the TE410. They work perfect. The only problem at the start was that we tried loading RedHat 9.0 them and not even the ethernet card worked. Upgrading to RH ES3.0 made things work flawlessly. We have similar config on 1850's but use whitebox instead of ES3.0. -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax pass through on te410p
We are using spandsp but find it unusable in a commercial environment, we are looking at changing to a dedicated hylafax server using an eicon diva PRI/E1-30 via asterisk.We know the server on it's own is a reliable configour only uncertainty is how good Asterisk is at handling pass throughfax on a te410p. Has anybody got good/bad experience with similar setup? Br/Kevin Brennan ___ 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] UK ISDN with Asterisk
Hi, Get yourself a 'Fritz!Card PCI' - also marketed by BT themselves as a 'BT Speedway ISDN' adapter - these seem to be the most cheap and supported of low-end ISDN2 adapters Will do - they seem pretty inexpensive (even for the BT Speedway card is only about £35). From doing a bit of poking, SuSE 9.1 seems to be the latest OS for which drivers are available. Is anyone using one of these cards successfully, and if so, on SuSE? One more question (and probably a pretty basic one, but i'm not that familiar with PSTNs) - will i need two of these cards in order to use both channels? Looking forward to getting this going now, and much more confident, thanks for your support! Henry. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
Yes the digium cards are relatively cheap compared to traditional telephony cards. A four port Eicon BRI card costs as much as the digium 4 port E1 so on a per channel basis (8 vs 120) the digium is very reasonable. Must think in terms of bang for buck before opening mouth next time. As for the server, well it is a case of the appropriate tool for the job, there is no SLA requirement for redundant power supply, raid, multiple CPU's, uber expensive reg ECC ram etc for the local servers in this particular application. The 750's are cheap enough to treat as disposable and I can get two of them for less than the cost of a redundant 1850, giving me a hot swap spare at each site. The central server will probably end up around the 5 digits mark and it'll have all the fruit, although personally I believe that reg ECC sdram is nothing more than a huge rort. Craig - Original Message - From: Max W Blackmer Jr [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 09, 2005 5:46 AM Subject: RE: [Asterisk-Users] Sangoma VS. Digium spending over $A10,000 in the process. The cards are more expensive than the server they're going into (Dell poweredge 750's). When a GPL'd hardware It is obvious that you have never experienced high end servers. We have had a single server cost as much as $20,000 and that is nothing but high performance hardware(Raid, REG ECC memory[mirrored for redundancy], Dual Xeon Server). Then you add in any specialized hardware that can easily up the cost to $30,000. and that is just one machine. when you need performance you pay for it one way or another. A lot of times it is better to pay more for reliability and performance. Max ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK ISDN with Asterisk
This would be a good solution but be aware that at this time the Fritz! may not handle DID (specifically PTP mode). The AVM drivers will not support DID. The mISDN drivers and fritz! cards do seem to handle DID but chan_capi doesn't pass the call to Asterisk (although you can see the call coming in with capi debug enabled). You might be able to get DID and frtiz! working with a combination of mISDN drivers (Kernel 2.6.9, 2.6.10 won't detect the fritz! card) and chan_mISDN. Craig - Original Message - From: Gavin Hamill [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, April 09, 2005 7:01 AM Subject: Re: [Asterisk-Users] UK ISDN with Asterisk On Friday 08 April 2005 23:33, Henry Owens wrote: Hi all, My question is: can Asterisk work well as a small office (8 extensions) PBX, with a mixture of analogue and IP phones, on an ISDN2e telephone line from BT? Sure, no problem at all.. Get yourself a 'Fritz!Card PCI' - also marketed by BT themselves as a 'BT Speedway ISDN' adapter - these seem to be the most cheap and supported of low-end ISDN2 adapters. chan_capi will deal with things like both B-channels so you can happily receive two calls on the same number, and deal with MSNs (Multiple Subscriber Numbers) gracefully since these are more likely on UK ISDN2e service than true DDIs. gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware dimesioning issues
Hello I am in the process of putting together a short term calling card solution that is rapidly deployable for charity events, and would apreciate some guidence on hardware dimensioning for the solution I have a test system running on an old P3 laptop, so in principle the solution works : It is configured as follows: Latest CVS of asterisk (well as of about 3 weeks ago) AreskiCC as the card solution Latest RPM of PostgreSQL Latest RPM of apache Latest RPM of php / pgphp 4 SIP accounts for the phones 1 SIP account with 4 concurrent calls for the lines Sipura 1001's as the ATA, DTMF phones on the end. It has a simple extension.conf User dials - runs DeadAGI(Areskicc.php) User goes on to enter PIN, phone number and then is connected (subject to credit and b-number being availible) The only difference between this test system and the production system is the number of lines. I need it to be able to run 80 extensions and therefor 80 lines (presented by SIP) How large should the processor, memory etc be - could anyone suggest a Dell / similar system that would be good for our needs. I don't need any zaptel hardware, as the places this is going to (its intended to be movable - not mobile per-se but movable) will only have outside internet connections, a local SIP provider is helping us which is why its SIP both sides. Thank you for your time on this matter David ___ 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] UK ISDN with Asterisk
Hi, Thanks for the tip - is there a better ISDN card (i don't mind paying extra) for compatibility with Asterisk? Is there any Digium hardware that will do what i need to do? I'm basically looking for a really reliable solution, with (relatively) easy setup and good compatibility, and don't mind paying more for it. -- Henry. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax pass through on te410p
If your hardware isn't getting clean data to spandsp, why should it be able to get clean data to a hylafax box? Unless you fix the config problem that stops spandsp working, there is no reason to expect a pass-through to a modem bank and hylafax to work. Regards, Steve Kevin Brennan wrote: We are using spandsp but find it unusable in a commercial environment, we are looking at changing to a dedicated hylafax server using an eicon diva PRI/E1-30 via asterisk. We know the server on it's own is a reliable config our only uncertainty is how good Asterisk is at handling pass through fax on a te410p. Has anybody got good/bad experience with similar setup ? Br/Kevin Brennan ___ 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] HOW TO SET THE TIME TO DIAL AFTER astcc-accountnum and astcc-phonenum
when a call comes the astcc-accountnum plays and ask the caller about the card number and after playing astcc-accountnum a period of time is given for the caller to dial his card number but the problem here is the short of the time given ,and i dont know where and how can i setup the time. __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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] UK ISDN with Asterisk
Henry Owens wrote: Hi, Get yourself a 'Fritz!Card PCI' - also marketed by BT themselves as a 'BT Speedway ISDN' adapter - these seem to be the most cheap and supported of low-end ISDN2 adapters Will do - they seem pretty inexpensive (even for the BT Speedway card is only about £35). From doing a bit of poking, SuSE 9.1 seems to be the latest OS for which drivers are available. Is anyone using one of these cards successfully, and if so, on SuSE? One more question (and probably a pretty basic one, but i'm not that familiar with PSTNs) - will i need two of these cards in order to use both channels? Looking forward to getting this going now, and much more confident, thanks for your support! Henry, hi; I'm running a small Asterisk PABX under SuSE 9.1. I have one analogue (PSTN) line, a single ISDN2e connection (i.e. 2 channels - one adaptor card) and a London PSTN number which gets routed to me via IAX, and I support 2 internal SIP phones and 4 internal analogue handsets. DID and whatever CLID is called in ISDN work fine. CLI on the analogue line is a nightmare because the Digium hardware doesn't support BT's CLI, so I have a modem picking that up and inserting it into Asterisk with (so far) variable results. Outgoing calls go either via the landlines, or via the Docklands-terminated IAX channel. All works pretty well - looks like just the sort of solution your client may need. Do be aware that supporting multiple ISDN2e cards might problematic. Not impossible, but problematic... jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ 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] UK ISDN with Asterisk
John, Thanks very much for the detailed response, that sounds pretty much like what i'm looking for (1x BT ISDN2e and 1x analogue). Are you using one of the Digium 4 port BRI cards, or what hardware are you using? It would be my intention to use the ISDN primarily for incoming, and VoIP for outgoing to cut costs, and increase functionality. You mentioned your PSTN number is routed to you via IAX; can that number be included in local directories? I don't think the analogue CLI should be a problem, since the ISDN should be taking most of the incoming calls. Does CLI work ok on the ISDN? At this point i would intend to use only 1 ISDN card, so i'll cross the multiple card bridge when (and if) i come to it. Thanks again! Henry. John Daragon wrote: Henry Owens wrote: Hi, Get yourself a 'Fritz!Card PCI' - also marketed by BT themselves as a 'BT Speedway ISDN' adapter - these seem to be the most cheap and supported of low-end ISDN2 adapters Will do - they seem pretty inexpensive (even for the BT Speedway card is only about £35). From doing a bit of poking, SuSE 9.1 seems to be the latest OS for which drivers are available. Is anyone using one of these cards successfully, and if so, on SuSE? One more question (and probably a pretty basic one, but i'm not that familiar with PSTNs) - will i need two of these cards in order to use both channels? Looking forward to getting this going now, and much more confident, thanks for your support! Henry, hi; I'm running a small Asterisk PABX under SuSE 9.1. I have one analogue (PSTN) line, a single ISDN2e connection (i.e. 2 channels - one adaptor card) and a London PSTN number which gets routed to me via IAX, and I support 2 internal SIP phones and 4 internal analogue handsets. DID and whatever CLID is called in ISDN work fine. CLI on the analogue line is a nightmare because the Digium hardware doesn't support BT's CLI, so I have a modem picking that up and inserting it into Asterisk with (so far) variable results. Outgoing calls go either via the landlines, or via the Docklands-terminated IAX channel. All works pretty well - looks like just the sort of solution your client may need. Do be aware that supporting multiple ISDN2e cards might problematic. Not impossible, but problematic... jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] unlimited iax termination
Serves you right for offering a bait and switch deal. If you are selling unlimited that's what it should be. Why would you be surprised if someone wants to use the unlimited feature? What's wrong with selling a 1000 minutes for $10 plan? I guess you are afraid someone will then offer an unlimited plan and take all the business! So you all offer unlimited, even though you can't deliver it and hide the real details in the fine print. So much for truth in marketing. There's laws to protect us from this kind of marketing, it's a shame they aren't used more often. Unfortunately, the marketing profession (world wide) has gotten to the point of how can we stretch the wording to influence a buy decision without outright lying. I'd swear a prereq for filling any marketing position is for one to have experience selling used cars. Read the fine print for... - satellite TV (HD, first months different rate, termination fee) - car leases (front-end and back-end fees, milage limits) - telephone company (home vs business line cost) - breakfast food - cisco phones (actual cost for a new working legal sip phone) - cellular usage plans - attorney's fees - TV ads (the first 100 callers get two for the price of one) - stock brokers (influenced by back door commissions) - unlimited voip plans (411 1-900 costs, international calls) It's become common practice to state one thing and place limits (or conditions) on that statement within the fine print. And, our legal system(s) seem to support that fine-print approach. The unlimited plans (regardless of which itsp) are no different. Technical folks are educated to define things in a clear concise manner, black or white, if... then..., spend time to find and remove exceptions to any spec, works or doesn't work, etc. Marketing is almost the complete opposite, and none of us (as individuals) can change that with the exception of individually electing not to buy. Your line drawn in the sand will likely be at a different location then the next person's. So, it's simply buyer be ware! ___ 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] How many FXS/FXO ports do I need?
I'm new to phone systems and phone wiring and I couldn't find an answer to this question on the wiki or google. My understanding is that a standard residential/business phone line carries the signal over 2 wires. Your 4-wire RJ11 wiring supports 2 phone lines. Given that each line takes 2 wires, and there are 8 wires in an FXO port, can I conceivably support 4 phone lines on one FXO port? On the phone/FXS side of things, can you also have multiple lines per FXS port? If I want to hookup 5 phones to my residential phone service with 2 lines, what # of FXO FXS ports do I need? Thanks for your clarification... Others have already addressed most of your questions. You have lots of different choices on how you address 5 phones and 2 lines. A couple choices include: - use the digium TDM card with two fxo modules (to connect to the two pstn lines), and one fxs module (to ring all of the five phones) - use two Sipura spa-3000 adapters (each adapter can support one pstn line and one fxs port, and you decide whether the two fxs ports provided on the adapters have one or more of the five phones attached to them. eg, business vs home phones. - replace all of your analog phones with voip sip phones (and ethernet wiring). No need for fxs ports. I kind of like the spa-3000 approach since those adapters allow you to make a decision on how you want your system to function, while also allowing you to change your mind and support your phones in a different way at some later date. Lots of little features built into those boxes. If you use the TDM card approach, any time your asterisk system is down (for any reason), your phones are all down. That will likely be a problem for you as you learn how to do things with asterisk, and some of those things require you to stop asterisk restart it. If you use the spa-3000 approach, you can configure the boxes to have all incoming pstn calls ring through to your phones (even when asterisk is down, or AC power is down). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax pass through on te410p
Ok - point taken - but we're running Asterisk as a SIP/PSTN gateway and we don't seem to have any other noticable problems, ok fax is more sensitive. We've tried different versions of spandsp and it does not fix anything, ok perhaps this shows problem is not spandsp - so where/how to start looking for a fix - any pointers anyone. If your hardware isn't getting clean data to spandsp, why should it be able to get clean data to a hylafax box? Unless you fix the config problem that stops spandsp working, there is no reason to expect a pass-through to a modem bank and hylafax to work. Regards, Steve Kevin Brennan wrote: We are using spandsp but find it unusable in a commercial environment, we are looking at changing to a dedicated hylafax server using an eicon diva PRI/E1-30 via asterisk. We know the server on it's own is a reliable config our only uncertainty is how good Asterisk is at handling pass through fax on a te410p. Has anybody got good/bad experience with similar setup ? Br/Kevin Brennan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP peer doesn't report busy properly
Hi Remco, -Original Message- I'm using wengo for my outgoing calls (SIP). However, whenever a number is busy, asterisk plays a message that the number you dialed is not available instead of a busy signal. How can I get the 'normal' PSTN tones (like number not in use tone or busy tone etc) Or is this not possible with SIP? Yes, it is possible to signal busy. However, the signalling must be correct through the entire path, so that also includes a requirement for your provider (Wengo). You can start by checking what signal they return to you by checking the DIALSTATUS variable after a Dial. Best regards, Florian ___ 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] Call rejected by XXX: No authority found
Everyone, I beg pardon to probably demand help of what had discussed many times, earlier. But I really stuck and earlier replies couldn't help me out. My first szenario connects two servers via IAX2. One is static IP the second is a nated dnyamic host. I could register the dynamic host succesfully at the static one. Routing calls to it with my dialplan gets denied/rejected due to missing authority on the remote side. I REALLY put this up several times different as regarded by many examples from voip.org. Still it doesn't work. Please anyone give me hints where I could find some useful information. Alex Fitterling My second -- Handyrechnung zu hoch? Tipp: SMS und MMS mit GMX Seien Sie so frei: Alle Infos unter http://www.gmx.net/de/go/freesms ___ 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] dyndns alias clients: needs to register in iax.conf as well?
One important question i ask my self is whether my asterisk server (it uses nat, which in public uses a dns alias as well), needs to register itself (with the register statement in iax.conf) at a host not behind a router? Would this be mandatory in any case asterisk is behind a router, or can I setup the peer using host = dnsalias as well, even if I use NAT? Regards, A.Fitterling -- Handyrechnung zu hoch? Tipp: SMS und MMS mit GMX Seien Sie so frei: Alle Infos unter http://www.gmx.net/de/go/freesms ___ 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 phone extensions at a remote site
I am in the proscess of integrating a clients remote and head office phone systems. Currenty each office has their own PBX and trunk lines. I am recommending that they put in an Asterisk server at the Head office with a WAN link to the remote office and switch to IP phones. Trunk lines at the remote site would be returned to the TELCO. External calls over the PSTN from the remote office would be routed over the WAN to the head office and through Asterisk to the PSTN trunk lines. All phones would then become extensions (both remote and head office locations). I want Person A in the remote office to dial an extension number and get Person B in the head office. What I am unsure about is if person A and Person B are both at the remote site and Asterisk PBX is at the head office, can A and B talk directly to each pther without traversing the WAN link? Has anyone done this before? What is the quality of the call if they have? Any information is useful. begin:vcard fn:Carey Mould n:Mould;Carey org:E2 Systems Limited adr:237 Old hope Road;;Suite 11 12, technology Innovation Centre;Kingston;;Kgn 6;Jamaica email;internet:[EMAIL PROTECTED] title:CEO/Consultant tel;work:(876) 512-2680 x-mozilla-html:FALSE url:http://www.e2team.com version:2.1 end:vcard ___ 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] Shorewall settings?
I use following settings in shorewall: (for connections established to the firewall) ACCEPT netfwudp 4569,5060,1:2 (all outgoing connections are permitted) Someone, please, comment on that to attest! I appreciate... A.Fittering -- Handyrechnung zu hoch? Tipp: SMS und MMS mit GMX Seien Sie so frei: Alle Infos unter http://www.gmx.net/de/go/freesms ___ 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] oh323 on @homeasterisk
Can you please detail the steps you have taken to successfully compile this on @home asterisk? Regards Mike - Original Message - From: CM Rahman Jr. [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, April 09, 2005 4:09 PM Subject: [Asterisk-Users] oh323 on @homeasterisk Anybody here added oh323 to @homeasterisk? I have compiled and add the oh323. I am wondering if anybody able to add the oh323 under web interface AMP? If anybody did it or know how to do it, please let me know. It has option for sip, IAX.. why not add h323 !! Thanks ** C.M. Rahman Jr. ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.5 - Release Date: 7/04/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Memory Requirements
MemTotal: 2074808 kB MemFree:417420 kB Buffers: 39396 kB Cached:1547124 kB SwapCached: 0 kB Active: 471180 kB Inactive: 1131508 kB HighTotal: 1179392 kB HighFree: 233536 kB LowTotal: 895416 kB LowFree:183884 kB SwapTotal: 2031608 kB SwapFree: 2031368 kB Dirty: 332 kB Writeback: 0 kB Mapped: 37696 kB Slab:43616 kB Committed_AS: 126244 kB PageTables: 1192 kB VmallocTotal: 106488 kB VmallocUsed: 3104 kB VmallocChunk: 103104 kB HugePages_Total: 0 HugePages_Free: 0 Hugepagesize: 2048 kB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cameron Schaus Sent: Saturday, April 09, 2005 1:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Memory Requirements On Fri, Apr 08, 2005 at 07:01:08PM -0500, Eric Rees wrote: I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB of memory. This is serving about 75 sip clients, Polycom500's and 600's. We are running into problems with the memory. Asterisk, right now, is using about 1.8GB of system memory. I am using Asterisk 1.0.7, Zaptel 1.0.7 with Digiums TE410 1xT1 RBS and 1xT1 PRI, Libpri 1.0.7 on Fedora Core 3. My question; is this normal or do I need more memory or is there a more serious underlying problem. How are measuring Asterisk memory usage? You're not counting the memory consumed by the filesystem cache, are you? Cam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK ISDN with Asterisk
Henry Owens wrote: John, Thanks very much for the detailed response, that sounds pretty much like what i'm looking for (1x BT ISDN2e and 1x analogue). Are you using one of the Digium 4 port BRI cards, or what hardware are you using? I'm using an AVM Fritz card with chan_capi. They're pretty cheap on eBay if you're suffering sticker shock... Of course, they're not as efficient as the active ones, but they're a lot cheaper and you already own the PC, I guess. It would be my intention to use the ISDN primarily for incoming, and VoIP for outgoing to cut costs, and increase functionality. You mentioned your PSTN number is routed to you via IAX; can that number be included in local directories? H... Probably not. I'm using vioptalk.org's Prepay Silver which allocates a geographic number. But it's *voiptalk's* number as far as the network is concerned, so I've no idea how you'd get it into a directory. I chose this because, although the company is ex-directory, I want people to be able to phone back so I'll show the geographic number in outgoing CLI. Some providers allow you to specify your own CLI on outgoing SIP, if that's any use. I don't think the analogue CLI should be a problem, since the ISDN should be taking most of the incoming calls. Does CLI work ok on the ISDN? Oh, yes. Of course, it's not *quite* the same. Here's an example : Analogue CLI :01460 234068 ISDN CLI*:441460234068 So if you want to call people back, you're going to have to play with extensions.conf... * Yes, I *know* that's not what it's really called... At this point i would intend to use only 1 ISDN card, so i'll cross the multiple card bridge when (and if) i come to it. OK. Drop me an email if you need any help (bearing in mind that paying clients get first dibs on my time!). jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Memory Requirements
Eric Rees wrote: MemTotal: 2074808 kB MemFree:417420 kB Buffers: 39396 kB Cached:1547124 kB SwapCached: 0 kB Active: 471180 kB That's a total memory usage for the entire OS of only 107MB: (Total-Free)-Cached. Tony ___ 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] Terrible crackling on analogue line and X100P card
Dear all ... I'm experiencing terrible trouble with crackling and noise on an analogue line connected to an X100P (compatible) card. I've checked the line with a normal analogue phone and it works fine, clear as a bell, but any outgoing or incoming calls to Asterisk are almost completely drowned out by loud crackling. I've attempted to adjust the RX and TX gains, but to no avail. There's also an echo, but only one way. I'm assuming this is a separate issue so I've not done much to investigate that, but I may be wrong so if it is related does anyone have any suggestions? I never had this trouble with ISDN, but then I wouldn't would I? :) If anyone can wave a magic wand, or at the very least point me to a website where I can get my own magic wand, please let me know. Thanks Stuart ___ 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] unlimited iax termination
Rich Adamson wrote: Serves you right for offering a bait and switch deal. If you are selling unlimited that's what it should be. Why would you be surprised if someone wants to use the unlimited feature? What's wrong with selling a 1000 minutes for $10 plan? I guess you are afraid someone will then offer an unlimited plan and take all the business! So you all offer unlimited, even though you can't deliver it and hide the real details in the fine print. So much for truth in marketing. There's laws to protect us from this kind of marketing, it's a shame they aren't used more often. Unfortunately, the marketing profession (world wide) has gotten to the point of how can we stretch the wording to influence a buy decision without outright lying. I'd swear a prereq for filling any marketing position is for one to have experience selling used cars. Read the fine print for... - satellite TV (HD, first months different rate, termination fee) - car leases (front-end and back-end fees, milage limits) - telephone company (home vs business line cost) - breakfast food - cisco phones (actual cost for a new working legal sip phone) - cellular usage plans - attorney's fees - TV ads (the first 100 callers get two for the price of one) - stock brokers (influenced by back door commissions) - unlimited voip plans (411 1-900 costs, international calls) It's become common practice to state one thing and place limits (or conditions) on that statement within the fine print. And, our legal system(s) seem to support that fine-print approach. Can we make a list of what is the limit for each provider? The unlimited plans (regardless of which itsp) are no different. Technical folks are educated to define things in a clear concise manner, black or white, if... then..., spend time to find and remove exceptions to any spec, works or doesn't work, etc. Marketing is almost the complete opposite, and none of us (as individuals) can change that with the exception of individually electing not to buy. Your line drawn in the sand will likely be at a different location then the next person's. So, it's simply buyer be ware! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax pass through on te410p
I had the same problem at one site. We could not receive faxes with spandsp reliably. Our solution that seems to have worked with no problems so far was to use a SPA-2000 to a fax machine. - Original Message - From: Kevin Brennan [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 09, 2005 7:16 AM Subject: Re: [Asterisk-Users] fax pass through on te410p Ok - point taken - but we're running Asterisk as a SIP/PSTN gateway and we don't seem to have any other noticable problems, ok fax is more sensitive. We've tried different versions of spandsp and it does not fix anything, ok perhaps this shows problem is not spandsp - so where/how to start looking for a fix - any pointers anyone. If your hardware isn't getting clean data to spandsp, why should it be able to get clean data to a hylafax box? Unless you fix the config problem that stops spandsp working, there is no reason to expect a pass-through to a modem bank and hylafax to work. Regards, Steve Kevin Brennan wrote: We are using spandsp but find it unusable in a commercial environment, we are looking at changing to a dedicated hylafax server using an eicon diva PRI/E1-30 via asterisk. We know the server on it's own is a reliable config our only uncertainty is how good Asterisk is at handling pass through fax on a te410p. Has anybody got good/bad experience with similar setup ? Br/Kevin Brennan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] unlimited iax termination
Folks, Let's try trimming the replies. I'm sick of wading through 100 lines of reply to find a single line comment. Chris Mason www.anguillaguide.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] unlimited iax termination
I am trying to put together a matrix. Please send me links, corrections, additions, flames, etc. http://www.geekgazette.com/index.php?option=com_contenttask=viewid=25Item id=26 -Kerry ___ 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] Terrible crackling on analogue line and X100P card
It Stuart...Wonder if We're long lost cousins or something...Name here is Bill Ford... Anyway...It sounds like a mechanical problem. Maybe something as simple as dirty contacts on the RJ-11 on the X100P. You say you've checked the line...but have you replaced the cable from the demark to the serverYou might try pulling the X100 and reseating it, also clean the contacts on the card's rj. On Apr 9, 2005 8:56 AM, Stuart Ford [EMAIL PROTECTED] wrote: Dear all ... I'm experiencing terrible trouble with crackling and noise on an analogue line connected to an X100P (compatible) card. I've checked the line with a normal analogue phone and it works fine, clear as a bell, but any outgoing or incoming calls to Asterisk are almost completely drowned out by loud crackling. I've attempted to adjust the RX and TX gains, but to no avail. There's also an echo, but only one way. I'm assuming this is a separate issue so I've not done much to investigate that, but I may be wrong so if it is related does anyone have any suggestions? I never had this trouble with ISDN, but then I wouldn't would I? :) If anyone can wave a magic wand, or at the very least point me to a website where I can get my own magic wand, please let me know. Thanks Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to change language using manager interface?
How do I change the language when I do commands from the manager interface? It seems that if I originate a call to a mailbox it will always speak English. I have set the language to da in sip.conf general context, but it still speaks English. I have no problems when using a phone, everything is in Danish. Is there a manager interface command to change language? Thorben ___ 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] Dialing With Backgound Music
How can play music when is clients phone ringing in dial progress. ___ 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] Terrible crackling on analogue line and X100P card
I have a similar situation but it seems to vary from call to call sometimes. Using 2 digium genuine x100p's in a dell with riser card. I'm wondering if it is something to do with the riser because it doesn't seem to matter if I swap various cords, positions, etc. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stuart Ford Sent: Saturday, April 09, 2005 9:57 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Terrible crackling on analogue line and X100P card Dear all ... I'm experiencing terrible trouble with crackling and noise on an analogue line connected to an X100P (compatible) card. I've checked the line with a normal analogue phone and it works fine, clear as a bell, but any outgoing or incoming calls to Asterisk are almost completely drowned out by loud crackling. I've attempted to adjust the RX and TX gains, but to no avail. There's also an echo, but only one way. I'm assuming this is a separate issue so I've not done much to investigate that, but I may be wrong so if it is related does anyone have any suggestions? I never had this trouble with ISDN, but then I wouldn't would I? :) If anyone can wave a magic wand, or at the very least point me to a website where I can get my own magic wand, please let me know. Thanks Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing With Backgound Music
Dial(SIP/100,30,tm) On Apr 9, 2005 5:50 PM, Ugur GUNCER [EMAIL PROTECTED] wrote: How can play music when is clients phone ringing in dial progress. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Terrible crackling on analogue line and X100Pcard
Dean Collins wrote ... Using 2 digium genuine x100p's in a dell with riser card. I'm wondering if it is something to do with the riser because it doesn't seem to matter if I swap various cords, positions, etc. Right, that's interesting. My card too is in a Dell (2550) with a riser card. That's a pig! Has *anything* you've done improved it at all? Stuart ___ 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] Terrible crackling on analogue line and X100P card
I have a very similar problem that I have been grappling with for a while. I've got a genuine TDM400P with four FXS ports and am using an Eicon Server quad BRI ISDN (using CAPI) for external calls. To date we have had no luck at all in diagnosing this problem as we too have periodic problems where the crackling occurs only sometimes, but affects all calls that are in progress (including those using the TDM400P and SIP calls that do not). Asterisk does not report any problems when this problem occurs, but it is sufficiently bad to force everyone to terminate in-progress calls (at which time everything works fine again). Appreciate hearing if you guys find a resolution to the problem that you are having, as we have had zero luck so far. dean collins wrote: I have a similar situation but it seems to vary from call to call sometimes. Using 2 digium genuine x100p's in a dell with riser card. I'm wondering if it is something to do with the riser because it doesn't seem to matter if I swap various cords, positions, etc. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Stuart Ford Sent: Saturday, April 09, 2005 9:57 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Terrible crackling on analogue line and X100P card Dear all ... I'm experiencing terrible trouble with crackling and noise on an analogue line connected to an X100P (compatible) card. I've checked the line with a normal analogue phone and it works fine, clear as a bell, but any outgoing or incoming calls to Asterisk are almost completely drowned out by loud crackling. I've attempted to adjust the RX and TX gains, but to no avail. There's also an echo, but only one way. I'm assuming this is a separate issue so I've not done much to investigate that, but I may be wrong so if it is related does anyone have any suggestions? I never had this trouble with ISDN, but then I wouldn't would I? :) If anyone can wave a magic wand, or at the very least point me to a website where I can get my own magic wand, please let me know. Thanks Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to change language using manager interface?
Thorben Jensen a écrit : How do I change the language when I do commands from the manager interface? It seems that if I originate a call to a mailbox it will always speak English. I have set the language to da in sip.conf general context, but it still speaks English. I have no problems when using a phone, everything is in Danish. You have to use SetLanguage(da) in you dialplan. Is there a manager interface command to change language? Thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing With Backgound Music
Ugur GUNCER wrote: How can play music when is clients phone ringing in dial progress. Usually you read the documentation. At the Asterisk CLI do a show applications to show you what Asterisk apps are available. Also see musiconhold.conf.sample in the Asterisk source directory (in the configs directory). To see detailed help for a specific application, like Dial, do show application dial. Pay special attention to the m option to Dial. Don't worry about the t option at this time (and don't use that option). The t option is actually a good one. If someone tells you to use it, you can pretty much assume they are a newbie and should take their advice with a grain of salt. t and T are only for a SPECIFIC type of call transfer and most people don't need it. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] s extension doesn't work with ata
Drew Einhorn wrote: The ATA generates it's own dialtone, and waits for the user to dial a number, before sending anything to the * box. So one of the first examples in the in the Brief Introduction to Dialplans from Vol. 1 of the Asterisk Documentation Project. [incoming] exten = s,1,Answer() exten = s,2,Playback(goodbye) exten = s,3,Hangup() does not work. The ATA generates a Dialtone and waits for the user to dial, then as soon as the user presses some keys. The ATA sends that extension was not found in [incoming] This example is elaborated into a simple example IVR. But how do we get the intial prompt to play on an ATA? In MY extensions.conf I have a comment above [incoming] that says something like Calls without a destination number land here, usually from the PSTN. s is ONLY EVER called when Asterisk doesn't know what number was dialed. This (generally) only happens if a call is coming in on an ANALOG port, or if the call is coming in on a T-1/E-1 port that does not have DID/DDI service on it. An IP Phone or ATA normally send the number dialed to Asterisk and therefore if you dial 5551212 then the ATA will send the call to exten = 5551212,1,Blah( Now if your ATA is not sending the correct numbers or not waiting for you to finish dialing then the problem is with ATA and NOT Asterisk. You didn't bother to tell us what ATA you are using, so I can't really give you any more advice. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Running a Marco from the dial command
[marco-voicerec]
exten = s,1,noop(${ARG1})
exten = s,2,Background(custom/recordwarn)
A nice thought, to name macros for Mark, marco. Won't work in the
dialplan though.
Also, *8 is usually used for picking up a ringing phone. See features.conf.
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Re: [Asterisk-Users] Call rejected by XXX: No authority found
My first szenario connects two servers via IAX2. One is static IP the second is a nated dnyamic host. I could register the dynamic host succesfully at the static one. Routing calls to it with my dialplan gets denied/rejected due to missing authority on the remote side. I REALLY put this up several times different as regarded by many examples from voip.org. Still it doesn't work. Please anyone give me hints where I could find some useful information. Looking at CLI of the asterisk behing NAT, do you see anything trying to authenticate when you call from another box? How would the asterisk behind NAT ever know anyone was calling it without forwarding port 4569 to it (or at least allowing traffic inside on it)? One of the boxes has to begin a dialogue, and that would be the one behind NAT. I'm not betting my life on the above, it's just a free opinion which I'm sure 10,000 people will jump in and correct if necessary. ___ 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] Terrible crackling on analogue line and X100P card
Forgot to mention - we are using an IBM xSeries 206 Server, so the Dell riser card may not be the issue if we are having the same problem. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Damian Funnell wrote: I have a very similar problem that I have been grappling with for a while. I've got a genuine TDM400P with four FXS ports and am using an Eicon Server quad BRI ISDN (using CAPI) for external calls. To date we have had no luck at all in diagnosing this problem as we too have periodic problems where the crackling occurs only sometimes, but affects all calls that are in progress (including those using the TDM400P and SIP calls that do not). Asterisk does not report any problems when this problem occurs, but it is sufficiently bad to force everyone to terminate in-progress calls (at which time everything works fine again). Appreciate hearing if you guys find a resolution to the problem that you are having, as we have had zero luck so far. dean collins wrote: I have a similar situation but it seems to vary from call to call sometimes. Using 2 digium genuine x100p's in a dell with riser card. I'm wondering if it is something to do with the riser because it doesn't seem to matter if I swap various cords, positions, etc. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Stuart Ford Sent: Saturday, April 09, 2005 9:57 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Terrible crackling on analogue line and X100P card Dear all ... I'm experiencing terrible trouble with crackling and noise on an analogue line connected to an X100P (compatible) card. I've checked the line with a normal analogue phone and it works fine, clear as a bell, but any outgoing or incoming calls to Asterisk are almost completely drowned out by loud crackling. I've attempted to adjust the RX and TX gains, but to no avail. There's also an echo, but only one way. I'm assuming this is a separate issue so I've not done much to investigate that, but I may be wrong so if it is related does anyone have any suggestions? I never had this trouble with ISDN, but then I wouldn't would I? :) If anyone can wave a magic wand, or at the very least point me to a website where I can get my own magic wand, please let me know. Thanks Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unlimited iax termination
Kerry Garrison wrote: I am trying to put together a matrix. Please send me links, corrections, additions, flames, etc. http://www.geekgazette.com/index.php?option=com_contenttask=viewid=25Item id=26 -Kerry Kerry, you did a great job, ... (I made a bookmark of it!!!) However, I wanted to find the real definition of unlimited of each provider. BTW, NuFone is missing. A suggestion to the list: Can you add the web site, please? Again, a wonderful creation. I am sure when the providers will see it, they will try to get the best place, ;-) bye Ronald ___ 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] Running a Marco from the dial command
Oh my gosh! I've been staring so long at it that I didn't even see my
typo. I was not talking about *8.I am using the prefix of 8 instead of
9. Like 8401234.
Regards,
Chris
- Original Message -
From: Wilson Pickett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 09, 2005 11:49 AM
Subject: Re: [Asterisk-Users] Running a Marco from the dial command
[marco-voicerec]
exten = s,1,noop(${ARG1})
exten = s,2,Background(custom/recordwarn)
A nice thought, to name macros for Mark, marco. Won't work in the
dialplan though.
Also, *8 is usually used for picking up a ringing phone. See features.conf.
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[Asterisk-Users] Asterisk as protocol conventer beetwen SIP and H.323
Hello, have successfully installed Asterisk 1.o with H.323 driver and made configuration: GW (Hardware)- GnuGK - Asterisk and i call into asterisk from the PSTN network and it's work fine, but i need to make conversion from SIP small gateways to H.323. I need to make configuration like that: (Normal Phones - SIP Gateways -) x many - Asterisk - GnuGK (H.323) - Gateway (H.323) SIP Gateway and H.323 Gateway supports g.729 - i need the g729 codec into Asterisk? Can i mark sip gateways that i will can see on h.323 gateway witch from SIP gateway it comes? Can you write sample configs for me? Im Asterisk newbie :) Regards, Adam Rybak ___ 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] how to pass G723.1
Kamran Ahmad wrote: hello how to pass G723.1 to other side is there any softphone using g723.1. i want to use G723.1 in my voice communication. Microsoft Netmeeting can use G723.1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as protocol conventer beetwen SIP and H.323
Hi, Try the OH323 implementation, we found it works better. Everyone has different experiences oviously.. Cheers, Sahil On Sat, 9 Apr 2005, Adam Rybak wrote: Hello, have successfully installed Asterisk 1.o with H.323 driver and made configuration: GW (Hardware)- GnuGK - Asterisk and i call into asterisk from the PSTN network and it's work fine, but i need to make conversion from SIP small gateways to H.323. I need to make configuration like that: (Normal Phones - SIP Gateways -) x many - Asterisk - GnuGK (H.323) - Gateway (H.323) SIP Gateway and H.323 Gateway supports g.729 - i need the g729 codec into Asterisk? Can i mark sip gateways that i will can see on h.323 gateway witch from SIP gateway it comes? Can you write sample configs for me? Im Asterisk newbie :) Regards, Adam Rybak ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA and NAT traversal
I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and my * server. My SPA is behind a NAT accessing a server which is also behind a NAT but SIP and RTP ports are forwarded to it. My SPA can successfully register. It can call another extension which is inside the * local net and the inside phone can call the SPA. But, no speech path either way. I have NAT=YES and the two invite parameters are set to NO. Thoughts? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Dual Servers
Hi all, I am trying to set up two asterisk servers (SrvA and SrvB), and what I want to get done is that if I dial 1X on SrvB the call must be routed to extension X on SrvA and if I dial 2X on SrvA the call must be routed to extension X on SrvB. I've read the www.voip-info.org wiki abouta sterisk dual servers but couldn't succeed on get it working. Perhaps someone that has a working dialplan similar to what I want to do could post his config files or explain what to do. Thanks in advance. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DS3000P - 20 E1 capacity on single card
I just checked digium's site. Looks like next big thing is coming to town DS3 on single card. Would be nice to know how many channels it can handle. Anybody had his hands on this card or knows some details ? regards m. ___ 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] unlimited iax termination
Message: 11 Date: Sat, 9 Apr 2005 08:21:16 -0700 From: Kerry Garrison [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] unlimited iax termination To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I am trying to put together a matrix. Please send me links, corrections, additions, flames, etc. http://www.geekgazette.com/index.php?option=com_contenttask=viewid=25; Item id=26 -Kerry Great idea, I would like to see one for DID/800 incoming also. Jeff ___ 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] DS3000P - 20 E1 capacity on single card
izo wrote: I just checked digium's site. Looks like next big thing is coming to town DS3 on single card. Would be nice to know how many channels it can handle. Anybody had his hands on this card or knows some details ? Please God, if you can hear me, don't let them use a TigerJet chipet. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Hardware dimesioning issues
I sent this earlier today. I didn't see my copy of the mail arrive back. Does anyone know if I am supposed to get back any of my posts or is there a setting I need to change. If it has been reflected properly this morning, please accept my applogies for the re-send. David -- Hello I am in the process of putting together a short term calling card solution that is rapidly deployable for charity events, and would apreciate some guidence on hardware dimensioning for the solution I have a test system running on an old P3 laptop, so in principle the solution works : It is configured as follows: Latest CVS of asterisk (well as of about 3 weeks ago) AreskiCC as the card solution Latest RPM of PostgreSQL Latest RPM of apache Latest RPM of php / pgphp 4 SIP accounts for the phones 1 SIP account with 4 concurrent calls for the lines Sipura 1001's as the ATA, DTMF phones on the end. It has a simple extension.conf User dials - runs DeadAGI(Areskicc.php) User goes on to enter PIN, phone number and then is connected (subject to credit and b-number being availible) The only difference between this test system and the production system is the number of lines. I need it to be able to run 80 extensions and therefor 80 lines (presented by SIP) How large should the processor, memory etc be - could anyone suggest a Dell / similar system that would be good for our needs. I don't need any zaptel hardware, as the places this is going to (its intended to be movable - not mobile per-se but movable) will only have outside internet connections, a local SIP provider is helping us which is why its SIP both sides. Thank you for your time on this matter David ___ 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] SPA and NAT traversal
Jim Sturtevant wrote: I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and my * server. My SPA is behind a NAT accessing a server which is also behind a NAT but SIP and RTP ports are forwarded to it. My SPA can successfully register. It can call another extension which is inside the * local net and the inside phone can call the SPA. But, no speech path either way. I have NAT=YES and the two invite parameters are set to NO. I'm desperately trying to get your sip.conf file telepathically but all I'm getting is images from your Martha Stewart porn collection. *shudder* In addition to nat=yes you also need localnet= and externip= set, as shown in sip.conf.sample. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as protocol conventer beetwen SIP and H.323
Cytowanie Sahil Gupta [EMAIL PROTECTED]: [...] Hi, Try the OH323 implementation, we found it works better. Everyone has different experiences oviously.. Thanks, just compiled oh323 0.6.5. But still don't know how force asterisk to act as protocol converter. Regards, Adam ___ 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] Terrible crackling on analogue line and X100P card
I am currently trying to solve this problem aswell with a TDM400p card and going out the FXO port to the PSTN .. If anyone runs into a solution, would be great news. T On Sat, 9 Apr 2005, Stuart Ford wrote: Dear all ... I'm experiencing terrible trouble with crackling and noise on an analogue line connected to an X100P (compatible) card. I've checked the line with a normal analogue phone and it works fine, clear as a bell, but any outgoing or incoming calls to Asterisk are almost completely drowned out by loud crackling. I've attempted to adjust the RX and TX gains, but to no avail. There's also an echo, but only one way. I'm assuming this is a separate issue so I've not done much to investigate that, but I may be wrong so if it is related does anyone have any suggestions? I never had this trouble with ISDN, but then I wouldn't would I? :) If anyone can wave a magic wand, or at the very least point me to a website where I can get my own magic wand, please let me know. Thanks Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD no longer doing IAX?
Last night I signed up for a FWD account and was hoping to use iax to connect thier server. I have been unable to connect as of yet. I get a: Registration of '64' rejected: Registration Refused. I used the iax section of http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWDto try and help me get this going. iax2 show peers gives me. Name/Username Host Mask Port Statusfwd-gw/64 65.39.205.121 (S) 255.255.255.255 4569 OK (76 ms) I can login to my account via thier web page so it would seem that the account is set up. Here is the string I am using in iax.conf. register = 64:[EMAIL PROTECTED] I set off a message to thier tech support but I am just wondering if any of you were having any problems. -Scott ___ 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] CallerID name lookup AGI script
Hi all,
My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote
an AGI script that does the following:
1) If it's a toll free number (800|888|877|866), set the CallerID name to
TollFree Caller
2) Use curl to look up the number in Google phonebook
3) If a business listing, set the CallerID name to business name, as is.
4) If it's a residential listing, reverse the listing so it's last name first,
then set the CallerID name to that.
5) If there's no match in Google phonebook, look up the NPA/NXX on
www.areacodedownload.com and set the CallerID name to @ST RATECENTER where
ST is the two-letter state abbreviation, and RATECENTER is the name of
telco rate center in that state.
Thought some of you might find this AGI script useful, so I'm including it
below. It requires the Asterisk::AGI perl module.
There are other reverse phone lookup sources that are more complete than
Google's, but they are harder to screen scrape. Also, I probably could have
made this a little cleaner if I used the Google API rather than screen
scraping with curl/perl. Please feel free to take a shot at making any of
those modifications.
Here's a snippet from my extensions.conf where it gets called:
exten = s,1,AGI(callerid.agi|${CALLERIDNUM})
exten = s,2,SetCallerId,${googlename} ${CALLERIDNUM}
exten = s,3,Dial(${PHONES},30,r)
exten = s,4,Answer
exten = s,5,Wait(2)
exten = s,6,Voicemail(u3001)
exten = s,7,Hangup
And here's the script:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$number = $ARGV[0];
if ($number =~ m/(800|888|877|866)\d{7}/) {
$AGI-set_variable('googlename', \TollFree Caller\);
exit 0;
}
open(RESULTS, /usr/bin/curl -s -m 2 -A Mozilla/4.0 http://www.google.com/search
?q=phonebook:$number |);
while (RESULTS) {
if (m/Residential Phonebook/) {
$reverse = 1;
@fields = split(//);
}
if (m/Business Phonebook/) {
@fields = split(//);
}
if (m/did not match any/) {
@digits = split(//, $number);
$npa = $digits[0] . $digits[1] . $digits[2];
$nxx = $digits[3] . $digits[4] . $digits[5];
open(LOCATION, /usr/bin/curl -s -m 2 -A Mozilla/4.0 http://www.areacodedown
load.com/$npa/$nxx/ |);
while (LOCATION) {
if (m/State/) {
$line = LOCATION;
$line =~ m/\\#CACACA\\w* (\w\w)\/td/;
$name = [EMAIL PROTECTED];
}
if (m/Rate Center/) {
$line = LOCATION;
$line =~ m/\\#CACACA\((\w|\s)*)\/td/;
$name = $name . . $1;
}
}
$AGI-set_variable('googlename', \$name\);
exit 0;
}
}
@result = split(/-/, $fields[35]);
chop($result[0]);
if ($reverse) {
@words = split(/ /, $result[0]);
$last = pop(@words);
unshift(@words, $last,);
foreach $word (@words) {
$name = $name . $word . ;
}
}
if ($reverse == 0) {
$name = $result[0];
}
$AGI-set_variable('googlename', \$name\);
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[Asterisk-Users] AgentLogin to MeetMe conference?
How can I configure AgentLogin to connect the agent to a MeetMe conference? Or, can I achieve similar functionality through other means? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 ___ 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] How to change language using manager interface?
| | How do I change the language when I do commands from the manager | interface? | It seems that if I originate a call to a mailbox it will always speak | English. I have set the language to da in sip.conf general context, but | it | still speaks English. I have no problems when using a phone, everything | is | in Danish. | | | | You have to use SetLanguage(da) in you dialplan. | | Is there a manager interface command to change language? | | Thorben Hi, I am looking for a way to change language from the manager interface, I don't have a problem with the dialplan. Thorben ___ 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] FWD no longer doing IAX?
On Sat, 9 Apr 2005 11:57:20 -0700, Scott Wolfe wrote Last night I signed up for a FWD account and was hoping to use iax to connect thier server. I have been unable to connect as of yet. I get a: Registration of '64' rejected: Registration Refused. I used the iax section of http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD to try and help me get this going. iax2 show peers gives me. Name/Username Host Mask Port Status fwd-gw/64 65.39.205.121 (S) 255.255.255.255 4569 OK (76 ms) I can login to my account via thier web page so it would seem that the account is set up. Here is the string I am using in iax.conf. register = 64:[EMAIL PROTECTED] I set off a message to thier tech support but I am just wondering if any of you were having any problems. -Scott You have to activate IAX support by hand inside your FWD account. Go into their webpage and find the option to activate IAX, after you select it it should take about half an hour until you can use IAX. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ 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] DS3000P - 20 E1 capacity on single card
like it says, the equivalent of 20 E1's or 28 T1's and I guess you know how many channels a E1 or T1 PRI is On Sat, 9 Apr 2005, izo wrote: I just checked digium's site. Looks like next big thing is coming to town DS3 on single card. Would be nice to know how many channels it can handle. Anybody had his hands on this card or knows some details ? regards m. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA and NAT traversal
Thank you for your reply. There is a wealth of information on the wiki, etc. I turned on RTP debug and the SPA is not sending it's public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere... The SPA is behind a NAT and traversing the public IP network to get to the * server. It is successfully registering, thus I can ring a phone registered locally to the * server. I made sure localnet=192.168.2.9/255.255.255.0 (my local cfg for *) and externip=65.87.x.x (which is the public IP of my * server). The * server is behind a NAT as well with the 5060 and 16384-32767 UDP ports open. Based on RTP debug it appears the RTP packets are making it to the * server, the problem is the return address is the internal NAT address of the SPA 192.168.1.100 and not it's public address. Are you willing to share your Martha collection or are you going to keep it to yourself? :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Saturday, April 09, 2005 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA and NAT traversal Jim Sturtevant wrote: I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and my * server. My SPA is behind a NAT accessing a server which is also behind a NAT but SIP and RTP ports are forwarded to it. My SPA can successfully register. It can call another extension which is inside the * local net and the inside phone can call the SPA. But, no speech path either way. I have NAT=YES and the two invite parameters are set to NO. I'm desperately trying to get your sip.conf file telepathically but all I'm getting is images from your Martha Stewart porn collection. *shudder* In addition to nat=yes you also need localnet= and externip= set, as shown in sip.conf.sample. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD no longer doing IAX?
Hi, On Apr 9, 2005 2:57 PM, Scott Wolfe [EMAIL PROTECTED] wrote: I used the iax section of http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD to try and help me get this going. I followed the directions below, and things are still working. You must activate iax through fwd. Check this page: http://www.freeworlddialup.com/content/view/full/1501 I can login to my account via thier web page so it would seem that the account is set up. Here is the string I am using in iax.conf. register = 64:[EMAIL PROTECTED] register = 64:[EMAIL PROTECTED] My line is: register = 64:[EMAIL PROTECTED] iax2.fwdnet.net... Give the instructions at the above link a go and let me know if that works for you. Cheers ___ 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] SPA and NAT traversal
Jim Sturtevant wrote: Thank you for your reply. There is a wealth of information on the wiki, etc. I turned on RTP debug and the SPA is not sending it's public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere... nat=yes makes Asterisk use the public IP that is inserted by the far side NAT router instead of the private IP the SIP device puts in the packet. Perhaps there is a problem in your sip.conf that is causing the SPA's packets not to match anything. sip show peers will tell you if Asterisk is seeing the public or the private IP of the far end SPA. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] SPA and NAT traversal
Thank you for your reply. There is a wealth of information on the wiki, etc. I turned on RTP debug and the SPA is not sending it's public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere... Do I understand your question correctly: You have an SPA behind NAT1 and * and a second SIP device behind NAT2. Both devices register, but calls between the devices result in no audio? If that is the case, you can do one of two things: - set canreinvite=no for the devices' sip.conf entries, or - teach both devices to *stop* using their internal IPs for all communications and remove nat=yes from the entry for the SIP device inside NAT2. To set the SPA to give the correct IP, enable STUN, add a STUN server, and say Yes to Substitue VIA Addr. -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 ___ 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] OT: ManxPower 2005 European Tour
I've helped a lot of people on the mailing lists and on IRC #asterisk. and wanted to let people know that I will be in Europe between May 19 and June 21. Stockholm (VON 2005), Brussels (holiday/vacation), Amsterdam (holiday/vacation), and Madrid (Astricon). There are several weeks during my trip that I have no current plans for and may add other cities to my itinerary. I'm looking for recommendations for lodging and tourist activities in all of the above cities. I would be interested in meeting Asteriskers for drinks or coffee in any of these cities. I am also looking for employment in Europe. I would prefer the Benelux area, but all serious offers will be considered. I have experience in a number of areas including Asterisk/SIP/IAX (2 yrs), Linux (10 yrs), WAN/Frame/T-1/DSL (10 yrs), and more. I can do limited programming in C, Perl and PHP. I am a citizen of the USA and want to relocate to Europe. Eric Wieling [EMAIL PROTECTED] -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Syntax error near unexpected token 'fi'
Hi, During boot I am getting an error that says the following: Syntax error near unexpected token 'f'i' /etc/rc3.d/S09zaptel line 92 Any ideas what might be causing this? I am using Fedora 3 with latest Asterisk build Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Syntax error near unexpected token 'fi'
During boot I am getting an error that says the following: Syntax error near unexpected token 'f'i' /etc/rc3.d/S09zaptel line 92 Maybe you should look at line 92 in that file and see what's up with it? Or post it here... --Luki ___ 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] SPA and NAT traversal
In your second option using a STUN server would I need to setup my own STUN server? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Saturday, April 09, 2005 12:37 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SPA and NAT traversal Thank you for your reply. There is a wealth of information on the wiki, etc. I turned on RTP debug and the SPA is not sending it's public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere... Do I understand your question correctly: You have an SPA behind NAT1 and * and a second SIP device behind NAT2. Both devices register, but calls between the devices result in no audio? If that is the case, you can do one of two things: - set canreinvite=no for the devices' sip.conf entries, or - teach both devices to *stop* using their internal IPs for all communications and remove nat=yes from the entry for the SIP device inside NAT2. To set the SPA to give the correct IP, enable STUN, add a STUN server, and say Yes to Substitue VIA Addr. -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA and NAT traversal
In your second option using a STUN server would I need to setup my own STUN server? No, use FWD or xten's STUN servers. -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 ___ 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] s extension doesn't work with ata
On Apr 8, 2005, at 9:40 PM, Drew Einhorn wrote: ...But how do we get the intial prompt to play on an ATA? On many ATAs you can have it do a hot-line dial -- start a call when the phone is picked up. Perhaps you can have your ATA dial @servername (no phone number, just the @ sign and the server name). If you do this, this makes the phone act like a regular phone -- you pick up the phone and it is connected to the asterisk server, rather than the primary way most ATAs work -- you dial a number and it connects to the server after you are finished dialing. ___ 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] Grandstream HandyTone-488, * - FXO problems
Pardon my answering myself (and for the long post). But I do have it sort of working, and I come back with information on the GS HT-488, as well as questions related to SIP / DTMF issues. The GS HT-488 acts as a PSTN pass through device for 4 rings. If the phone attached to the FXS port hasn't picked up by 4 rings, it will by default answer, and you're at an internal (*) dial tone. You can also configure the HT-488 to dial a specific extention, which it will then do instead of dropping you at an internal dial tone. From there you can obviously do what ever you want with the call. (It would be nice if you could configure and/or disable the # rings before it switches over to VoIP. Maybe that will be something they will add to a firmware update someday.) For dialing out, you set up an extention for the FXO port, and dial that. It will ring once, and then present you with the PSTN line, dial tone and all. From there you (should be) are able to dial out. Now, here is my problem and question. Both the FXS and FXO ports are set up to use SIP INFO for DTMF. You would think that when you have dialed the FXO port, and are at the PSTN dial tone, the HT-488 will translate the SIP DTMF INFO passed through to the FXO port as audible DTMF on the PSTN line. This is not the case. So I really can't make outgoing calls yet. Now, I can change the FXS line to send DTMF in audio, which works, but I figure that sending DTMF in audio is not ideal. So I'm trying to translate the SIP DTMF INFO to DTMF in-audio. I've tried a few combinations of SipDTMFMode(inband) (trying to do a DTMF style translation, I guess), and Dial(SIP/gs1-FXO,10,D(PSTNnumber) ), but can't get it to work. Should I just suck it up and keep the FXS port using DTMF in-audio, or is there a way to get SIP DTMF INFO translated to DTMF tones in audio in the Dial settings for the FXO extension? Thanks! Dan Dan Perik wrote: I just got my shiny new Grandstream HandyTone-488 today. My goal is to use it to allow incoming/outgoing calls to PSTN using my normal ole' phone as usual. I will be switching over to using BroadVoice as my main phone #, but want that to be as seemless of a switchover as possible (for the wife and kids, and for people needing to call us). I've got the following working: FXS - * ( and then - BroadVoice ) ( BroadVoice - ) * - FXS FXO - * ( and then - FXS ) I don't have this working: ( FXS - ) * - FXO In other words, I can't seem to call out on my PSTN line from Asterisk. snip ___ 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] Netlogic inbound DID issue
I've seen this with @home.
Either trunk (under amp) and then dial(sip/trunk_name/extension)
or
Dial(IAX2/user_name:[EMAIL PROTECTED]/s)
James
On Fri, 18 Mar 2005 07:08:17 -0600, Matt Schulte [EMAIL PROTECTED]
wrote:
Per Mike's issue here, we're noticing this problem with older versions
of Asterisk (it would seem?), and especially distrib [EMAIL PROTECTED] As
he stated we're seeing 'No Authority Found' coming from the clients, in
[EMAIL PROTECTED] we get see the No Authority found on the server, and the
client
sees absolutely nothing.
What's strange is I personally run CVS-head at my house, dated 11/10/04,
it has no problems at all.
If anyone has info on this please help, it's killing us :D
Matt
-Original Message-
From: Mike Clark [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 17, 2005 11:02 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Netlogic inbound DID issue
Anyone out there using NetLogic DIDs? And have inbound working? I got
outbound working, but no joy so far with inbound. Here are the relevant
parts from my conf files:
iax.conf
[general]
tos=lowdelay
jitterbuffer=no
register = username:[EMAIL PROTECTED]
[netlogic]
type=friend
host=dynamic
context=sourcekit-main
auth=plaintext
username=
secret=
disallow=all
allow=ulaw
allow=all
extensions.conf
[sourcekit-sip]
exten = 101,1,Dial(SIP/SK-101,20)
exten = 101,2,Voicemail(u101)
exten = 101,102,Voicemail(b101)
exten = 101,103,Hangup
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
[sourcekit-main]
include=sourcekit-sip
exten = +19193233010,1,GoTo(sourcekit-sip,101,1)
exten = _1NXXNXX,1,SetCallerID(9193233010)
exten = _1NXXNXX,2,Dial(IAX2/netlogic/${EXTEN})
exten =
_1NXXNXX,3,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,4,Hangup
[netlogic]
include=sourcekit-main
and, thr debug output from * CLI:
Asterisk Ready.
*CLI iax2 debug
IAX2 Debugging Enabled
*CLI Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX
Subclass: NEW
Timestamp: 00017ms SCall: 00030 DCall: 0 [206.80.70.49:4569]
VERSION : 2
CALLED NUMBER : +19193233010
Unknown IE 045 : Present
CALLING NUMBER : +13362150564
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
LANGUAGE: en
CALLED CONTEXT : netlogic
USERNAME: username
FORMAT : 4
CAPABILITY : 2097151
ADSICPE : 2
DATE TIME : 175199382
Ignoring unknown information element 'Unknown IE' (45) of length 1 Mar
17 12:35:19 NOTICE[21100]: chan_iax2.c:5419 socket_read: Rejected
connect at
tempt from 206.80.70.49, who was trying to reach '[EMAIL PROTECTED]'
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REJECT
Timestamp: 00018ms SCall: 2 DCall: 00030 [206.80.70.49:4569]
CAUSE : No authority found
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--
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas 75503
903-793-1956
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Re: [Asterisk-Users] Terrible crackling on analogue line and X100P card
On Sat, 9 Apr 2005, Stuart Ford wrote: Dear all ... I'm experiencing terrible trouble with crackling and noise on an analogue line connected to an X100P (compatible) card. I've checked the line with a normal analogue phone and it works fine, clear as a bell, but any outgoing or incoming calls to Asterisk are almost completely drowned out by loud crackling. I've attempted to adjust the RX and TX gains, but to no avail. There's also an echo, but only one way. I'm assuming this is a separate issue so I've not done much to investigate that, but I may be wrong so if it is related does anyone have any suggestions? I never had this trouble with ISDN, but then I wouldn't would I? :) If anyone can wave a magic wand, or at the very least point me to a website where I can get my own magic wand, please let me know. Thanks Stuart Maybe a filter or isolation transformer is needed. Maybe both. Ask a ham radio enthusiast about such things. Maybe we have one reading this list? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any opinions on quality/service of Teliax?
I'm looking into the TelIAX pay-as-you-go plan. I'm assuming that they charge incoming calls minutes as well? Is there the $0.02 connection fee for the incoming call as well? Thanks, Jared From: [EMAIL PROTECTED] on behalf of Mohit Muthanna Sent: Fri 4/8/2005 12:30 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Any opinions on quality/service of Teliax? I've been using TelIAX for a while now. Outside of the occasional network hiccup, they certainly are one of the better providers out there. Their customer support is also very responsive. Highly recommended. Mohit. On Apr 8, 2005 12:32 PM, Brian McSpadden [EMAIL PROTECTED] wrote: On Apr 8, 2005 9:06 AM, Jacob Cazzell [EMAIL PROTECTED] wrote: Looking at alternative VoIP providers and I found Teliax. One of the features listed on their pay-as-you-go plan is unlimited incoming/outgoing connections. I am working on setting up a conference calling system for some of our traveling salepeople to call into for their weekly staff meetings. Right now our phone system limits the number of connected conf callers - this would be a perfect fit. There are so many VoIP providers out there, it's tough to know who's good and who's not. Any insight on Teliax is apprecaited! I also have been using them for a month or so. I'll have to say, they're great. Very responsive support, great selection of DIDs, and good voice quality. I'd definitely recommend them. Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] There are 10 types of people. Those who understand binary, and those who don't. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any opinions on quality/service of Teliax?
Yes and yes. On Apr 9, 2005 6:03 PM, Bellows, Jared [EMAIL PROTECTED] wrote: I'm looking into the TelIAX pay-as-you-go plan. I'm assuming that they charge incoming calls minutes as well? Is there the $0.02 connection fee for the incoming call as well? Thanks, Jared ___ 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] unlimited iax termination
Serves you right for offering a bait and switch deal. If you are selling unlimited that's what it should be. Why would you be surprised if someone wants to use the unlimited feature? What's wrong with selling a 1000 minutes for $10 plan? I guess you are afraid someone will then offer an unlimited plan and take all the business! So you all offer unlimited, even though you can't deliver it and hide the real details in the fine print. So much for truth in marketing. There's laws to protect us from this kind of marketing, it's a shame they aren't used more often. Unfortunately, the marketing profession (world wide) has gotten to the point of how can we stretch the wording to influence a buy decision without outright lying. I'd swear a prereq for filling any marketing position is for one to have experience selling used cars. Read the fine print for... - satellite TV (HD, first months different rate, termination fee) - car leases (front-end and back-end fees, milage limits) - telephone company (home vs business line cost) - breakfast food - cisco phones (actual cost for a new working legal sip phone) - cellular usage plans - attorney's fees - TV ads (the first 100 callers get two for the price of one) - stock brokers (influenced by back door commissions) - unlimited voip plans (411 1-900 costs, international calls) It's become common practice to state one thing and place limits (or conditions) on that statement within the fine print. And, our legal system(s) seem to support that fine-print approach. Can we make a list of what is the limit for each provider? Sure you can, in most cases. Just check the fine print in their service agreements, or whatever else they publish. If its not their, call them as a prospective customer. If they don't answer, then why bother to do business with them as that's going to be about the same level of service you get after signing up. ___ 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] unlimited iax termination
I am trying to put together a matrix. Please send me links, corrections, additions, flames, etc. http://www.geekgazette.com/index.php?option=com_contenttask=viewid=25Item id=26 Go look at the list on digium's site, free world dialup's site, the wiki, google, etc. ___ 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] Terrible crackling on analogue line and X100P card
I'm experiencing terrible trouble with crackling and noise on an analogue line connected to an X100P (compatible) card. I've checked the line with a normal analogue phone and it works fine, clear as a bell, but any outgoing or incoming calls to Asterisk are almost completely drowned out by loud crackling. I've attempted to adjust the RX and TX gains, but to no avail. There's also an echo, but only one way. I'm assuming this is a separate issue so I've not done much to investigate that, but I may be wrong so if it is related does anyone have any suggestions? I never had this trouble with ISDN, but then I wouldn't would I? :) If anyone can wave a magic wand, or at the very least point me to a website where I can get my own magic wand, please let me know. What country are you in, and does the chipset on the compat card support the telco standards in your country? If the chipset doesn't match your telco standards, there is a high probability you won't get rid of the echo. If it does match, then try echotraining=800 echocancel=yes Regarding the crackling noise, have you checked for shared interrupts (cat /proc/interrupts)? If you run cat /proc/interrupts every ten seconds, do you see calculated interrupt values of about 1,000? Go to /usr/src/zaptel directory and run ./zttest Do you get something close to 100% over some period of time? What version of asterisk are you running? ___ 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] unlimited iax termination
Rich Adamson wrote: Sure you can, in most cases. Just check the fine print in their service agreements, or whatever else they publish. If its not their, call them as a prospective customer. If they don't answer, then why bother to do business with them as that's going to be about the same level of service you get after signing up. You shouldn't have to call them. If you email them and always provide a phone number, you are allowing them to choose the most convenient way to answer your questions. If they are non-responsive you just include all the email inquiries somewhere within your website and let the viewer decide. If I read your six unanswered email inquiries to a vendor, all I need to see is that they were composed well and went unanswered. Very important to find out the minimum, increment and rounding methods used for billing calls. From what I have seen so far, it is usually not found on the website or faq when it really sucks. So email them and see how quick they respond when they know the answer is not the one you want to hear. ___ 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] Using manager interface to play aanouncments in aMeetMe
I've wondered about this as well. I suggest posting a bug to the bug tracker
and see if you can get a clarification or better yet, get someone to fix
this. It would be nice to override the clearing of the vars for Local
channels.
MATT---
-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED]
Sent: Friday, April 08, 2005 11:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin aMeetMe
A little more googling and wiki browsing shows that the default
behaviour
of the Local channel is to dump the variables. According to the wiki I
can append /n to the channel identifier to preserve variables, but this
does
not seem to be working. I'm running 1.0.7.
Can anyone lend me a clue?
Thanks,
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin
Sent: Friday, April 08, 2005 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin aMeetMe
I've run into a snag. I make extensive use of dynamic conferences
and thought it would be a no-brainer to pass the conference number
with the manager interface. Looking at the wiki and sample code,
I thought I had it right, but no joy.
PHP manager call-
$res = $as-connect();
if (!$res){ echo 'Error connection to the manager!'; exit();}
$res = $as-send_request('Originate', array('Channel' =
'Local/[EMAIL PROTECTED]', 'Context' = 'mm-announce', 'Exten' = '',
'Priority' = '1', 'Variable' = 'confNo=$confNo'));
$res = $as-disconnect();
extensions.conf-
[mm-announce]
exten = 9998,1,Answer
exten = 9998,2,noop,${confNo} ;to test, it was meetme(${confNo})
exten = 9998,3,Hangup
exten = ,1,Answer
exten = ,2,Playback(this-conf-will-end-in-5-minutes)
exten = ,3,Hangup
Console output-
== Manager 'MeetMe' logged on from 127.0.0.1
-- Executing Answer(Local/[EMAIL PROTECTED],2, ) in new
stack
-- Executing NoOp(Local/[EMAIL PROTECTED],2, ) in new stack
-- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new
stack
-- Executing Answer(Local/[EMAIL PROTECTED],1, ) in new
stack
-- Executing Playback(Local/[EMAIL PROTECTED],1,
this-conf-will-end-in-5-minutes) in new stack
-- Playing 'this-conf-will-end-in-5-minutes' (language 'en')
== Manager 'MeetMe' logged off from 127.0.0.1
So it appears that my variable ${confNo} is not being set, or at least
honored.
Any thoughts?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Thursday, April 07, 2005 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin a MeetMe
Hello,
Here's jsut a simple manager Action to send, make sure that you have an
extension set up to play the message(exten = 1234,1,Playback(file)) and
that's the extension that will be called from the meetme room. Also,
make
sure that that extension calls in to the meetme room extension with the
'q'
flag so that noone hears the welcome and leaving tone.
exten = 1234,1,Answer
exten = 1234,2,Playback(out_of_time)
exten = 1234,3,Hangup
Action: Originate
Channel: Local/[EMAIL PROTECTED]
Context: default
Exten: 1234
Priority: 1
where 78600051 is the exten to get to your meetme room.
Let me know if you have any questions,
MATT---
-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 7:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin a MeetMe
A sample would be great. I'm hoping that the Official MeetMe2
will have provisions for this, but until then I'll have a
fully functional scheduler.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Thursday, April 07, 2005 3:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Using manager interface to play
aanouncmentsin a MeetMe
just create an extension that plays the message and hangs up and use the
manager interface to drop it into the meetme room.
Let me know if you would like an example and I'll whip one up.
We do this kind of thing in astGUIclient to play DTMF tones
automatically in
meetme rooms.
MATT---
-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 6:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Using manager interface to play aanouncments
in a MeetMe
I am wrapping up a PHP addon script to my scheduling
framework and have it properly tracking and closing
conferences.
I need to play an announcement into the room that the
conference will end soon. I haven't found a great way
to do that. One way that I have thought of, but would
like to avoid is adding a Playback
Re: [Asterisk-Users] DS3000P - 20 E1 capacity on single card
On April 9, 2005 02:13 pm, Eric Wieling wrote: izo wrote: I just checked digium's site. Looks like next big thing is coming to town DS3 on single card. Would be nice to know how many channels it can handle. Anybody had his hands on this card or knows some details ? Please God, if you can hear me, don't let them use a TigerJet chipet. I don't think they will; their quad T1/E1/J1 have no such POS on them. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any opinions on quality/service of Teliax?
On Apr 9, 2005 5:03 PM, Bellows, Jared [EMAIL PROTECTED] wrote: I'm looking into the TelIAX pay-as-you-go plan. I'm assuming that they charge incoming calls minutes as well? Is there the $0.02 connection fee for the incoming call as well? That's the only thing they do that I could do without. But, for the service they provide, I'll gladly pay it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any opinions on quality/service of Teliax?
Brian McSpadden wrote: On Apr 9, 2005 5:03 PM, Bellows, Jared [EMAIL PROTECTED] wrote: I'm looking into the TelIAX pay-as-you-go plan. I'm assuming that they charge incoming calls minutes as well? Is there the $0.02 connection fee for the incoming call as well? That's the only thing they do that I could do without. But, for the service they provide, I'll gladly pay it. ___ With Teliax I noticed that the delay between the Dial command running and me hearing the ringback tone is unusually long. Not TERRIBLE, just unusual for a VoIP connection. More like the delay when dialing out of an analog port, but they don't use analog ports. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Terrible crackling on analogue line and X100Pcard
Rich Adamson wrote ... What country are you in, and does the chipset on the compat card support the telco standards in your country? I'm in the UK. The card was bought in the UK, but from Ebay, so I suppose it could have originated from anywhere. The card dials and answers calls without a problem, so it must be doing *something* right. I didn't *mean* to cheap out over this - I tried to buy a genuine Digium part, but they don't seem to do it any more and I can't find it for sale anywhere. The Ebay vendor claimed it was 100% compatible. The card reports itself as: 00:02.0 Communication controller: Individual Computers - Jens Schoenfeld Intel 537 When the wcfxo module loads, dmesg reports: Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 11 for device :00:02.0 Uhhuh. NMI received. Dazed and confused, but trying to continue You probably have a hardware problem with your RAM chips wcfxo: DAA mode is 'FCC' Found a Wildcard FXO: Generic Clone Registered tone zone 4 (United Kingdom) The 3rd and 4th lines are suspicous, but I've no idea what they mean. Does it refer to the system RAM or some sort of special RAM on the card? What is NMI? If the chipset doesn't match your telco standards, there is a high probability you won't get rid of the echo. If it does match, then try echotraining=800 echocancel=yes I already use those parameters in zapata.conf, they make no difference :( Regarding the crackling noise, have you checked for shared interrupts (cat /proc/interrupts)? This is the output: CPU0 0: 211266080 XT-PIC timer 2: 0 XT-PIC cascade 7: 488230 XT-PIC eth0 10:2113812 XT-PIC eth1 11: 211520617 XT-PIC aacraid, wcfxo 14: 11 XT-PIC ide0 NMI: 1 ERR: 60 It's sharing an interrupt with the RAID controller. I did try to separate the interrupts when I installed the card, but any combination other than that automatically assigned by the BIOS caused the Linux kernel to fail to even uncompress at boot time, much less boot the system, which struck me as a pretty alarming failure. If you run cat /proc/interrupts every ten seconds, do you see calculated interrupt values of about 1,000? I don't know what you mean here. Go to /usr/src/zaptel directory and run ./zttest Do you get something close to 100% over some period of time? Yep: # ./zttest Opened pseudo zap interface, measuring accuracy... 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% Is that good? What version of asterisk are you running? 1.0.7 plus Zaptel of the same version. Thanks Stuart ___ 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] DS3000P - 20 E1 capacity on single card
Andrew Kohlsmith wrote: On April 9, 2005 02:13 pm, Eric Wieling wrote: izo wrote: I just checked digium's site. Looks like next big thing is coming to town DS3 on single card. Would be nice to know how many channels it can handle. Anybody had his hands on this card or knows some details ? Please God, if you can hear me, don't let them use a TigerJet chipet. I don't think they will; their quad T1/E1/J1 have no such POS on them. Which specific Digium card does not use the TigerJet chip (as shown in lspci)? -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Configuring the Sipura for static IP and registering with Asterisk.
OK so now you have an IP address. Did you login and configure the Sipura? On Apr 7, 2005, at 1:04 AM, Rich Adamson wrote: I wish to configure my Sipura with static IP. I have set the static IP, but there is registration failure on doing so. Could you please tell me how do I go about configuring my Sipura for static IP and register it successfully with the Asterisk server. A few of the spa changes require the box be rebooted. Did you do that? Can you ping the sipura's ip address? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I set queue not to hangup?
I'm aware of the context=menu feature in queue.conf. This feature only works while the caller is waiting for an agent. What I want to do is allow the caller to press * during the conversation with the agent and exit the queue application without hanging up. On Mon, 4 Apr 2005, Richard Lyman wrote: Steve Edwards wrote: How can I configure queue() so that it does not hang up if the caller presses * to exit the queue? I want to continue the call so the caller can choose other services. allow the agent to be able to transfer, then create an exten in that context that does what you want. so the agent then presses '#xxx' (xxx = some other exten within that context) then the agent goes right back in the queue, and the caller goes on to the dialplan logic of that exten. ___ 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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 ___ 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] DS3000P - 20 E1 capacity on single card
On April 9, 2005 08:25 pm, Eric Wieling wrote: Which specific Digium card does not use the TigerJet chip (as shown in lspci)? TE405P: 05:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev 01) I imagine the TE410 and TE110 are both also similarly lspci'd. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank replacement
I enjoy using the Adit 600 with the new FXS cards via the controller T1 interfaces. Works well. I do have concerns with using the CMG card via MGCP. Has anyone done this? How is it working? On Apr 8, 2005, at 12:50 PM, Matt Schulte wrote: Word of warning, get the version 5 or higher FXS cards with the ADIT600, else you will have echo problems. This is just from personal experience. Supposedly the 5 and higher cards have dynamic impedance adjustment, it's worth it. Matt -Original Message- From: Peter Hoppe [mailto:[EMAIL PROTECTED] Sent: Friday, April 08, 2005 12:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Channel bank replacement Thank you so much for your answers already, I really appreciate it! I have looked into using an Adtran Total Access 750 platform instead, but got away from that idea after I saw the totally confusing amount of options of different modules I can buy. The Adit 600 seemed so much simpler to put together. Also, the Adit 600 had such an excellent appraisal in the asterisk voip-info - see http://www.voip-info.org/tiki-index.php?page=Asterisk%20Channel%20Bank But maybe I need to come back to the Adtran TA750. Unfortunately that platform seems to only offer 24 fxs ports per unit and I need to buy an expensive T1 card. I would buy the Digium T1 card - it seems that it is by far the least expensive card, but $500 is still something. That's why I toyed with the Adit 600 plus cmg card - all I need is a standard network card on the Asterisk machine. We have sorely abandoned the idea of using an extensive amount of voip phones on the property, as we are not a homogenous office setup (ppl also live on the property). This solution would mean * putting in an entire new cat5 network. I would be the person who would have to put it all in place - When would I be finished? In 2 years? 4 years? 10 years? * lots of admin hassle to enable all the phones / add new phones / remove phones * users can't easily extend stations at end points. With two wire phone they simply switch one parallel to the existing one - no admin hassle / extra hubs etc. * two wire technology enables us to buy almost any phone available. * security concerns with the SIP protocol. See http://secunia.com/advisories/8169/ as an example * users potentially plugging their laptops into the voip sockets and browsing/downloading away = lots of setup/admin hassle with the firewall (how do you block Kazaa?) * Phones potentially breaking when users unplug power during firmware download. For example, this is an issue with the Grandstream phone. The only alternative that seems feasible at the moment would be * a different channel bank than the adit 600 or * a voip gateway that multiplexes many fxs ports into one ethernet connection. But before I would go down that route I would have to be absolutely sure that the SIP conforms to the standard, the upgrades are free and the fxs ports are compatible with uk standard two wire phones. I found that some two wire phones actually use 4 wires - confusing * a bank of ATAs (handytone 286 or similar). I *really* don't like that solution, as it is a bad botch job and throws lots of issues like which REN they have, many power supplies (or one big one). I really ought to be red in the face for even mentioning that solution. But if nothing else is available, I would probably have to buy them in bulk, take the boards out and mount them in a 19'' box together with a hub so I build my own voip gateway :) maybe it's not so botch after all :) ) For connection to the PSTN: We have three BT lines, and again, we would not like to move over to a different technology like ISDN. The lines work for us, and 'if it ain't broke, don't fix it'. We would use three Sipura SPA-3000 interfaces to connect them to the internal network. The SPA-3000 is sold in the UK and has the CE approval, so it should legally be ok. I am experimenting with one unit at the moment, and am smacked by the literally hundreds of options it has. But I heard good reports about that one, so I expect it to work well in our setting. Hi Peter, I'm not sure how you are getting PSTN lines into your * box, but if it's not ISDN30, you might want to consider some of the cheap IAX phones on the market now rather than trying to soldier on with old analogue kit? e.g. http://www.iaxtalk.com/product_info.php?cPath=1products_id=29 Shipping for 30 units and UK power supplies was $340, and with the weak dollar right now, that works out at just over 40 quid per phone - I'm sure there's movement on the unit price when buying in bulk... Now remove the need for an Asterisk Quad-E1 / T1 interface card and you've dropped the cost by nearly a grand food for thought :) They also sell a single-ethernet-port version of the phone for $10 less if you have enough ethernet sockets. Cheers, Gavin. I got an Adtran 600 with 12 X FXO and 12 X FXS cards for $495 from Penny Doyen [EMAIL PROTECTED] With the strength of the
