Re: [Asterisk-Users] Re: Sangoma A102 cards testing
On Sun, 13 Feb 2005 22:36:45 +0100, Michiel van Baak [EMAIL PROTECTED] wrote: On 21:47, Sun 13 Feb 05, Vikram Rangnekar wrote: +++ Michael Devenijn [13/02/05 18:23 +0100]: Actually I am using a supermicro board the P4SCI wonder if I can turn off hyperthreading i dont think there is a bio option i'm running kernel 2.4.29 does it use hyperthreading and can i turn it off ? Kernel 2.4 does not have HT support. I beg to differ. Since at least kernel 2.4.26 (probably earlier) the standard Linux kernel does support HT and turns it on by default, as long as it's compiled with CONFIG_MPENTIUM4=y; I am sure because I have a machine running this way right now. You can check by running: cat /proc/cpuinfo It will list info for CPU 0 only. It's OK to check for HT this way: /proc/cpuinfo will report two cpu's for every ht-enabled processor you have on board. If it's indeed running with HT and you need to disable it, try noht on the kernel boot line; if your kernel doesn't support it, you will need to recompile it with CONFIG_M686 instead of CONFIG_MPENTIUM4. As this is way off-topic, if you have further questions please fell free to email me directly, it will be a pleasure to help. Best Regards, -- Durval Menezes (durval AT tmp DOT com DOT br, http://www.tmp.com.br/) ___ 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: card dialer phone
Rob at draughon.org writes I recently obtained a Western Electric multi-line phone and am seeking help with getting this beast working with *. The interesting stuff in my * implementation consists of a T100P card, a TDM400P card, and an Adtran TA750 channel bank with three quad-port FXS modules and a quad-port FXO. The TA750 is wired to a 24-port Cat 5 patch panel via a 25-pair Amp cable. The phone is a model 2662A1M; it has five lines, a hold button (I presume), card dialer capability, and a 25-pair Amp cable for connecting to The Phone System. (The card dialer feature, IMHO, scores major geek points. If you're not familiar with it, you take a special plastic card about the size of a credit card and punch out two tiny discs for each digit in a phone number. When it's time to call that number, you insert the card in the phone, take the handset off hook, push the START button, and--voila!--the phone speed dials your party.) Jerry has already posted the basics -- this is a traditional fat wire key system phone and will work with either 1A1 or 1A2 key equipment. The first pair is tip and ring, the second pair is A-lead control, the third is for the light. When you press the hold key, the A-lead is disconnected first; when you release the hold key, the line button pops out which disconnects tip and ring. The key telephone unit (KTU) card in the key service unit (KSU) chassis detects this sequence and puts a holding bridge on the line and changes the lamp from steady to winking. There is a group of telephone collectors putting together Asterisk boxes who will be able to fill you in on all relevant details, see www.ckts.info and join the list at http://lists.ckts.info/mailman/listinfo/voip ___ 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] chan_capi or chan_mISDN vs bristuff
On Fri, 11 Feb 2005, Peer Oliver Schmidt wrote: Remco Barende wrote: I'm currently using a HFC-S card for my ISDN BRI line with bristuff. The instability is driving me crazy however. [..] I have three different locations with HFC cards. I had the same stability problems on ALL of the installations. Since RC5 plus the florz patch *ALL* of the stability problems have vanished. No more seconds of silence, no more unavailability messages. It just works now. I won't touch the installations for a long time :-) I applied the florz patch but the problems remain. Now I get all sorts of weird errors on the console and I cannot make outgoing calls. Any clue what these errors are? Ri = 44651 TEI msg = 3 TEI = 7f Ri = 3800 TEI msg = 3 TEI = 7f Ri = 42399 TEI msg = 3 TEI = 7f Ri = 42409 TEI msg = 3 TEI = 7f Ri = 22078 TEI msg = 3 TEI = 7f Ri = 991 TEI msg = 3 TEI = 7f Feb 14 10:30:25 NOTICE[14777]: app_dial.c:762 dial_exec: Unable to create channel of type 'Zap' Ri = 36942 TEI msg = 3 TEI = 7f Feb 14 10:31:13 WARNING[14777]: pbx.c:444 pbx_exec: Stack overflow, cannot create another stack Ri = 25084 TEI msg = 3 TEI = 7f Feb 14 10:34:21 WARNING[14777]: chan_zap.c:7411 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 received TEI check request for TEI = 102 Feb 14 10:39:01 WARNING[14777]: chan_zap.c:7411 zt_pri_error: PRI: !! Not good - head of queue has not been transmitted yet Feb 14 10:39:33 WARNING[14777]: chan_zap.c:7411 zt_pri_error: PRI: !! Got reject for frame 8, but we have nothing -- resetting! ___ 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] spandsp asterisk 3/5
Good day all I want to know with version of spandsp works well with ether asterisk 1.0.3 or 1.0.5 Thanks Altus ___ 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 - SER Configuration
Alberto Zuin wrote: Yes, but I have to configure a route for each host in every host! A the moment i have about 120 Asterisk hosts and every astersk have about 50-100 users! Is for that I want a single sip proxy that route dial. I read more about ser, and the suggestion is to use ser for accounting and route, and asterisk only for PBX gateway and for voicemail. In my situation this isn't perfect because I have to use asterisk for sip login... What you do in this situation: Remember the point that Asterisk is a UA, not a proxy. You get Asterisk to register to SER with a particular account. When one of the other boxes dials [EMAIL PROTECTED] the request travels to Asterisk which dials a number on the SER box ([EMAIL PROTECTED]). SER looks in it's routing table to see where a.com is, and redirects the request there. Once the request gets to the Asterisk box at a.com, the Asterisk server checks the account name that the request is for and forwards it to the user. With record routing obviously the 100 - Trying, 180 - Ringing and 200 ok pass through all of the previous servers. This allows you to keep control of accounting etc at any box along the way. (I.E. one of your rules in SER might say that if a call is to a [EMAIL PROTECTED] then pass it to a PSTN gateway). With the record routing on, you would still get a message saying that the call had hung up even if you are not one end of the call. Your best bet would be to read up on some of the SIP documentation on the iptel.org site (particularly the introduction to SIP and the SER user's guide). Hope my ramblings make sense! :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Initializing two ISDN cards in isdn4linux
On Sat, 2005-02-12 at 12:20 +, JunkMail wrote: For the single card I was using with isdntool for initialization, wich works fine but has no support for two cards. Can anyone tell me exactly how to initialize the ISDN system manually ??? It all starts with modprobe -v hisax type=21,21 (loading hisax and telling it that we'll use two teles pci cards) and then ? what else ??? Not sure if this will help you - I ponce played with a mixture of single port cards... Can't remember where I got the 'id=' bits from.. # For one eicon PCI #modprobe hisax type=11 # For two eicon cards modprobe hisax type=11,11 id=201%202 # For Asuscom ISA # isapnp /etc/isapnp.conf #modprobe hisax type=12 irq=3 io=0x0100 -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN zaphfc - What kernel are you using successfully?
Hi, some people report good success with the zaphfc cards, others, incl. myself have mixed results. I am using the debian stock kernel 2.4.27 with mixed results. Anyone care to tell what kernel(s) you on successful zaphfc integrations? Thanks. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ 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] ATA's
On Sun, Feb 13, 2005 at 07:43:06PM -0600, Matthew Boehm wrote: We have had a big success with the Linksys PAP2-NA. 2 FX ports and 1 WAN port. Only downside is that only 1 call can be using 729 at a time. This has been confirmed with Linksys. They will be releasing PAP2-NAv2 in March to overcome this. In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. Personally, I dislike the lack of LEDs on the 2100. My 2100s have 3 LEDs, plus 2 for each RJ-45 port. Instead of just 2 for the SPA-2000. -- Nicolas Bougues ___ 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] ATA's
On Sun, Feb 13, 2005 at 10:39:36AM -0800, Luki wrote: The Sipuras have a ton of configurable parameters. If you understand them (and there is no good manual, unfortunately) then you can be of great benefit. Otherwise they'll be worthless. I particularly miss the dial-plan, distinctive ring and audio gain options on the Grandstreams. Remote syslog can also be useful for debugging. It all depends what you need, I guess. Further, the Sipuras have a more detailed status, that is accessible WHILE you are engaged in a conversation. I think you're paying a bit more for the 1000 (1 line version) as compared to the Grandstream 286, but if you need/want two independent lines, then the Spa 2000 is more economical (as Peter said). The Sipuras are really a dream to manage, particularly in an international environment. You can customize the tones, the rings, the voltages, the dialplan, the features... well, everything. They are (securely) remote manageable and upgradeable. They are rock solid. Sipura support is helpful in case you need them for complex issues. Voice quality is top notch. The Grandstreams are less manageable, have less parameters, have only american tones, no dialplan support, no auto-upgrade (well, they recently added some kind of support). Voice quality is OK. -- Nicolas Bougues ___ 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] Linphone / Kphone
Hi, I have * working with X-Lite and Sipura adapters, but I have one person who is linux based, and is trying to use Linphone and Kphone. His end works, but I get very bad echo on my end. Have any of you folks been able to get linux based soft phones working well with *? I'd appreciate links to howtos/docs if you have them, and/or samples of working configs for * and the linux softphones. Thanks much for sharing your experience. Darren ___ 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] Linphone / Kphone
On Mon, February 14, 2005 22:22, Darren Ellis said: I'd appreciate links to howtos/docs if you have them, and/or samples of working configs for * and the linux softphones. I gave up trying to use linux soft clients they all seem to have some fatal flaws or issues I could never fully get rid of and ended just using xten lite under wine... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ 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] Linphone / Kphone
Darren Ellis wrote: Hi, I have * working with X-Lite and Sipura adapters, but I have one person who is linux based, and is trying to use Linphone and Kphone. His end works, but I get very bad echo on my end. Have any of you folks been able to get linux based soft phones working well with *? If you get bad echo on your end, this is prolly acoustic echo (speaker feeding back into the mike) from his side. Try having him use a headset/mike combo or something and make another test. You'll see it will work much better... In general with softphones, you don't want to use the built-in microphone/speakers on any pc, unless you put the volume so low it won't feed the microphone again. The echo you experience, is your voice going out his speaker, feeding into his microphone again and then coming back to you. That's why it's only experienced at one side. Johan I'd appreciate links to howtos/docs if you have them, and/or samples of working configs for * and the linux softphones. Thanks much for sharing your experience. Darren ___ 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] Sipura 841 and paging function
I was browsing through the web config of a Sipura SPA-841 (Firmware 2.0.13) and noticed a setting marked 'paging' under supplementary services on the Phone settings page on the advanced admin login. Anyone know how it might be used? Could it be like the Snom - exten = 10,1,SetVar(VXML_URL=intercom=true) exten = 10,2,Dial(SIP/testuser) Craig ___ 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] speech recognition
hi i am looking for some info for speech recognition for example when someone call to my house asterisk ask for who hi want to call and he say the name david or susan (wife) or daniela etc... thanks 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] speech recognition
David D. Faerman wrote: hi i am looking for some info for speech recognition for example when someone call to my house asterisk ask for who hi want to call and he say the name david or susan (wife) or daniela etc... And the wife asks Who's Daniela? ;-) -- _/_/_/_/ _/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/ _/ Bill Maidment Maidment Enterprises Pty Ltd Unless you are named Alfred E. Newman, you may read only the odd numbered words (every other word beginning with the first) of the message above. If you have violated that, then you hereby owe the sender AU$10 for each even numbered word you have read. Adapted from Stupid Email Disclaimers (see http://www.goldmark.org/jeff/stupid-disclaimers/) ___ 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] Linphone / Kphone
On Mon, 2005-02-14 at 12:22, Darren Ellis wrote: Hi, I have * working with X-Lite and Sipura adapters, but I have one person who is linux based, and is trying to use Linphone and Kphone. His end works, but I get very bad echo on my end. Have any of you folks been able to get linux based soft phones working well with *? I'd appreciate links to howtos/docs if you have them, and/or samples of working configs for * and the linux softphones. Thanks much for sharing your experience. Darren Hi Darren, I have been using kphone 4.1.0 for a while with good results. The problem I had with echo, was because of the headset/ear-piece I was using. I now use a very cheap headset from Trust http://trust.com/products/productpics.aspx?artnr=13585 and there is no echo in either ends.. -Tor. ___ 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] equipament for use with Asterisk (call id and db access)
Dear friends, I need to make a software for a listen service. A room with 6 persons, 6 lines and 6 extensions. When a people (client) call for this room (external calls), depending of number, asterisk access a data base searching for that number and forwarding (propably whith a PABX) to a available people in room, with some informations about de client (identified by the number). Which equipaments i need to make this? any PABX? TDM card? how many FX0? Only this? Thanks very much in advance! Pablo Fernandes ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error: Unknown RTP codec 72 received
Hi all, I have setup two X-Lite phones and an Asterisk box. They are all on the same LAN and have private IP addresses assigned to them. I am able to place a call from either phone but the moment it is picked up (trying to be answered), it goes dead - as in no sound! I get the error, Unknown RTP codec 72 received. How can I go about this? Thanks, -- Rgds, Julius Kidubuka. When you do the common things in life in an uncommon way, you will command the attention of the world! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linphone / Kphone
Am Montag 14 Februar 2005 12:57 schrieb Tor Setane: On Mon, 2005-02-14 at 12:22, Darren Ellis wrote: Hi, I have * working with X-Lite and Sipura adapters, but I have one person who is linux based, and is trying to use Linphone and Kphone. His end works, but I get very bad echo on my end. Have any of you folks been able to get linux based soft phones working well with *? I'd appreciate links to howtos/docs if you have them, and/or samples of working configs for * and the linux softphones. Thanks much for sharing your experience. Maybe you wanna check out the softphone zip4x5 made by Zultys. It's the software which is used by the same hardphone. It has a lot of features and is much better than kphone and linphone. Jens ___ 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: Is there a Caller ID issue in the latest CVSStable
(Intentional top-post, due to relative brevity of answer) The error is a typo in the latest chan_sip.c in Stable. See my note on Mantis bug #3557 (softins). To fix, find line 3673 and change ast_isphonenumber(l) to !ast_isphonenumber(l) CVS HEAD does not have the typo, so is OK. Cheers Tony Robert L Mathews [EMAIL PROTECTED] wrote: Nicol?s Gudi?o [EMAIL PROTECTED] wrote: Paul, 1.0.5 stable suffers from caller id issues as well, at least for SIP channels. What fixed things for me was swapping in app_dial.c from 1.0.2 stable (didn't try others). You could also just diff app_dial.c between versions to find the problem but I took the lazy way out the first time around. Drumkilla reverted the callerid changes on the latest stable (thanks Russell!). You will be fine if you checkout stable from CVS now. Hmmm; I think I'm still having problems with it, using a completely fresh checkout and compile: Connected to Asterisk CVS-v1-0-02/11/05-17:34:08 I have two Zap FXS lines and two SIP phones, and: - Zap channel to Zap channel, caller ID works (displays correctly on the analog phone display). - SIP phone to Zap channel, caller ID works. - SIP phone to ZIP phone, caller ID does NOT work (Grandstream phone displays Err). - Zap channel to SIP phone, caller ID does NOT work. - Incoming Free World Dialup calls to Zap channel extension, caller ID works. - Incoming Free World Dialup calls to SIP phone extension, caller ID does NOT work. So it seems that asterisk stable, as of today, does not send correct caller ID on calls that end up on SIP phones, unless I'm doing something boneheaded (although I used almost-identical config files on 1.0.2 with no trouble). A tcpdump shows that asterisk is sending this in the SIP INVITE header to the phone: From: asterisk sip:[EMAIL PROTECTED]; (IP address obscured; it's correct in the original.) But somehow asterisk appears instead of the correct caller ID. Wasn't that the bug other people were seeing that the stable update was supposed to fix? Have I missed something obvious? -- Robert L Mathews, Tiger Technologies http://www.tigertech.net/ -- 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] OT: Aastra 390 - weird problem
On February 14, 2005 01:18 am, Matt Gibson wrote: It can receive calls both when receiving power, and when not receiving power. It can make calls only when not receiving power from the wall. I tried unplugging it for a good 10-15 minutes to make sure it was off for sufficient time, but still didn't make a difference. What happens when you plug it directly into the POTS line from the telco -- does it fail to send good DTMF when powered as well? -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
[Asterisk-Users] Error: Unknown RTP codec 72 received???
Hi all, I have setup two X-Lite phones and an Asterisk box. They are all on the same LAN and have private IP addresses assigned to them. I am able to place a call from either phone but the moment it is picked up (trying to be answered), it goes dead - as in no sound! I get two errors, Unknown RTP codec 72 received and RFC3389 support incomplete. How can I go about this? Thanks, -- Rgds, Julius Kidubuka. When you do the common things in life in an uncommon way, you will command the attention of the world! -- Rgds, Julius Kidubuka. When you do the common things in life in an uncommon way, you will command the attention of the world! -- Rgds, Julius Kidubuka. When you do the common things in life in an uncommon way, you will command the attention of the world! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 7, Issue 202
On Mon, Feb 14, Craig Guy wrote: I was browsing through the web config of a Sipura SPA-841 (Firmware 2.0.13) and noticed a setting marked 'paging' under supplementary services on the Phone settings page on the advanced admin login. Anyone know how it might be used? Could it be like the Snom - http://voip-info.org/wiki-SPA-841 Geoff ___ 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] speech recognition
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi David D. Faerman wrote: | hi i am looking for some info for speech recognition for example | when someone call to my house asterisk ask for who hi want to call | and he say the name david or susan (wife) or daniela etc... | Why not the easy way ? "Press 1 for Susan", "Press 2 for David", "Press 3 for Sam the Dog", "Press 4 for Nemo the Little Fish", "Press 5 to leave a message", "Press 6 to Hangup". rgds Joo Amaro | thanks 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 | | -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCEJ1SJUm/Bor63CERAvLdAJ9U0nKUzxFy/azVbe/ZgtDQ/WiKCQCgk247 EOJGYXBusZBxL94Pj/Pw/HU= =hVFw -END PGP SIGNATURE- ___ 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] Bristuff and Realtime
Hi, I would like to use Realtime extentions with a four bri card, the classic quodbri. Normally with that card I would use * bristuffed from Klaus-Peter Junghanns, but since that package is based on stable version there is no Realtime at all in it (I suppose). Did you knoww if someone has done a merger, or can help me in such task ? Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Reccomendation for reliable handsets
Thanks Mark I am definitely interested in the budgetone 102 but am a little concerned about the 10mbit only Ethernet ports !! From what I have read, these are relatively new models and I like the addition of a second port to daisy chain your PC from the same network connection, however why 10mbits and not 100mbits ??, I would have thought this would be a minimum these days, I don't know anyone who still runs 10mbits to the desktop, and im not too happy about bottlenecking my customers fast Ethernet network with these phones A real shame really. Does anybody know if Grandstream will be addressing this or indeed if they have any current models with at least 100mbit ports Regards -Original Message- From: Mark Benson [mailto:[EMAIL PROTECTED] Sent: 03 February 2005 14:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Reccomendation for reliable handsets I have been using an IN1002 generic handset (supposed to be an unbranded cisco copy but I am skeptical) for a few months (6months+) now, and it seems pretty stable - however I haven't found a reliable supplier Also there is almost no support for them.. I have switched to the grandstream budgetone 102 and they seem pretty good too. You can pretty much plug in and forget it with both phones. They do lock up occasionally (once a month to once every 3 months). I have yet to upgrade the firmware on the grandstreams... Mark Brett, Gary wrote: Sorry to move this up the list again, but does anybody have any advice on this -Original Message- From: Brett, Gary [mailto:[EMAIL PROTECTED] Sent: 02 February 2005 10:49 To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] Reccomendation for reliable handsets Hi there I'm sure this question has been raised a number of times before, but unfortunately I do not have direct access to the archives I am about to roll out Asterisk to a few companies and would like to hear your experiences about the various handsets/phones that are Asterisk compatible I am primarily looking for 2 options, the first being a cheaper model which will provide reliability whilst still maintaining a reasonable feature set, and a reliable model from the more expensive range with more features But the definite focus here is on reliability and ease of maintenance Any help or advice would be greatly appreciated; I would really like to hear your experiences/recommendations Cheers Gary ___ 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] ISDN zaphfc - What kernel are you using successfully?
Thibault Lamy wrote: some people report good success with the zaphfc cards, others, incl. myself have mixed results. I am using the debian stock kernel 2.4.27 with mixed results. We are using 2.6.10 self-built kernel on debian unstable zapfhc works fine, we are able to send/receive calls and calelrid works What version asterisk and bristuff do you use? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ 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] Digium Cards connecting to BT
Hi there Just a general question, has anybody experienced any problems with any Digium telephony cards in the UK, specifically with BT (British Telecom) lines. I just want to make sure there are no compatibility issues before purchasing cards, (mainly TDM400P's) Any comments would be greatly appreciated 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] asterisk in New-Zealand
Good day all Anyone doing asterisk in New-Zealand? ___ 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 in New-Zealand
Altus Snyman wrote: Good day all Anyone doing asterisk in New-Zealand? But of course! The Daily Asterisk News is run out of New Zealand! We are also local distributor for Digium gear. We provide all of the support for products also. Let us know if you have any questions etc. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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 in Singapore.
In the vain of asterisk in new-zealand... Anyone know of a reliable source of digium gear in singapore? Also where to pick up IP phones, anyone any clues? Ta Jonathan signature.asc Description: This is a digitally signed message part ___ 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 in Singapore.
I can get you a good deal if you import the from South-Africa..Let me know.Altus On Mon, 2005-02-14 at 15:38, Jonathan Gill wrote: In the vain of asterisk in new-zealand... Anyone know of a reliable source of digium gear in singapore? Also where to pick up IP phones, anyone any clues? Ta Jonathan __ ___ 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] SIP configurations
Hello, I wanna configure Asterisk to work with iptel.org proxy. I have already created an account in iptel.org; what steps should I do?. I want to test the configurations using X-Lite and some help to configure it out could be nice too. Thx -- -DdC ___ 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] speech recognition
I am not much into speech recognition, but I know that a major company only had success when they simplified the menus so as to only ask simple yes/no-questions in this manner: Do you have problems with your internet connection? (yes = Do you have a black modem?) (no = Do you have problems with your telephone?) In sequence you would be guided through simple yes/no questions. It works like a charm. ___ 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 in Singapore.
Hi Altus What sort of price are you able to get? Im only looking for prob 2 (cheap) ip phones right now, maybe more later if all goes well... And as this is personal stuff, im on a tight budget. Ta Jonathan On Mon, 2005-02-14 at 15:40 +0200, Altus Snyman wrote: I can get you a good deal if you import the from South-Africa..Let me know.Altus On Mon, 2005-02-14 at 15:38, Jonathan Gill wrote: In the vain of asterisk in new-zealand... Anyone know of a reliable source of digium gear in singapore? Also where to pick up IP phones, anyone any clues? Ta Jonathan __ ___ 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 signature.asc Description: This is a digitally signed message part ___ 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] Reccomendation for reliable handsets
On Monday 14 February 2005 13:00, Brett, Gary wrote: Thanks Mark I am definitely interested in the budgetone 102 but am a little concerned about the 10mbit only Ethernet ports !! From what I have read, these are relatively new models and I like the addition of a second port to daisy chain your PC from the same network connection, however why 10mbits and not 100mbits ??, I would have thought this would be a minimum these days, I don't know anyone who still runs 10mbits to the desktop, and im not too happy about bottlenecking my customers fast Ethernet network with these phones A real shame really. Does anybody know if Grandstream will be addressing this or indeed if they have any current models with at least 100mbit ports Please do not top post. I don't think there is a single IP phone which can flood a 10Mbps port. You do not need 100Mbps on a phone unless it has a passthrough to a PC. Let's see, using a 64Kbps codec and being generous, will use 100Kbps on the wire. Assuming the 10Mbps port can reliably run at 8Mbps, that means the phone would have to have 8 * 1000 / 100 = 80 concurrent RTP streams. Can anyone see anything wrong with my rough calculations? B -Original Message- From: Mark Benson [mailto:[EMAIL PROTECTED] Sent: 03 February 2005 14:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Reccomendation for reliable handsets I have been using an IN1002 generic handset (supposed to be an unbranded cisco copy but I am skeptical) for a few months (6months+) now, and it seems pretty stable - however I haven't found a reliable supplier Also there is almost no support for them.. I have switched to the grandstream budgetone 102 and they seem pretty good too. You can pretty much plug in and forget it with both phones. They do lock up occasionally (once a month to once every 3 months). I have yet to upgrade the firmware on the grandstreams... Mark Brett, Gary wrote: Sorry to move this up the list again, but does anybody have any advice on this -Original Message- From: Brett, Gary [mailto:[EMAIL PROTECTED] Sent: 02 February 2005 10:49 To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] Reccomendation for reliable handsets Hi there I'm sure this question has been raised a number of times before, but unfortunately I do not have direct access to the archives I am about to roll out Asterisk to a few companies and would like to hear your experiences about the various handsets/phones that are Asterisk compatible I am primarily looking for 2 options, the first being a cheaper model which will provide reliability whilst still maintaining a reasonable feature set, and a reliable model from the more expensive range with more features But the definite focus here is on reliability and ease of maintenance Any help or advice would be greatly appreciated; I would really like to hear your experiences/recommendations Cheers Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Digium Cards connecting to BT
Hi, There are several people on the UK mailing list (I am one) that have purchased the TDM400P FXO and are having problems with disconnect. Basically the cards are great (sound quality etc) but give some issues with detecting a UK remote hang-up. Mainly an issue within IVR, MeetMe and VM. There are several of us trying to get to the bottom of this, either with fixes or workarounds. If you only want a couple of lines and ISDN isn't an option perhaps look at the Sipura 3000 they have one FXO and one FXS interface. Also they don't cost the earth are UK approved, and available in the UK so no import duty. Regards, Chris -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: 14 February 2005 13:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Digium Cards connecting to BT Hi there Just a general question, has anybody experienced any problems with any Digium telephony cards in the UK, specifically with BT (British Telecom) lines. I just want to make sure there are no compatibility issues before purchasing cards, (mainly TDM400P's) Any comments would be greatly appreciated 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 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] Reccomendation for reliable handsets
On Mon, 14 Feb 2005 14:11:15 + Bob Goddard [EMAIL PROTECTED] wrote: On Monday 14 February 2005 13:00, Brett, Gary wrote: Thanks Mark I am definitely interested in the budgetone 102 but am a little concerned about the 10mbit only Ethernet ports !! From what I have read, these are relatively new models and I like the addition of a second port to daisy chain your PC from the same network connection, however why 10mbits and not 100mbits ??, I would have thought this would be a minimum these days, I don't know anyone who still runs 10mbits to the desktop, and im not too happy about bottlenecking my customers fast Ethernet network with these phones A real shame really. Does anybody know if Grandstream will be addressing this or indeed if they have any current models with at least 100mbit ports Please do not top post. I don't think there is a single IP phone which can flood a 10Mbps port. You do not need 100Mbps on a phone unless it has a passthrough to a PC. Let's see, using a 64Kbps codec and being generous, will use 100Kbps on the wire. Assuming the 10Mbps port can reliably run at 8Mbps, that means the phone would have to have 8 * 1000 / 100 = 80 concurrent RTP streams. Can anyone see anything wrong with my rough calculations? B -Original Message- From: Mark Benson [mailto:[EMAIL PROTECTED] Sent: 03 February 2005 14:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Reccomendation for reliable handsets I have been using an IN1002 generic handset (supposed to be an unbranded cisco copy but I am skeptical) for a few months (6months+) now, and it seems pretty stable - however I haven't found a reliable supplier Also there is almost no support for them.. I have switched to the grandstream budgetone 102 and they seem pretty good too. You can pretty much plug in and forget it with both phones. They do lock up occasionally (once a month to once every 3 months). I have yet to upgrade the firmware on the grandstreams... Mark Brett, Gary wrote: Sorry to move this up the list again, but does anybody have any advice on this -Original Message- From: Brett, Gary [mailto:[EMAIL PROTECTED] Sent: 02 February 2005 10:49 To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] Reccomendation for reliable handsets Hi there I'm sure this question has been raised a number of times before, but unfortunately I do not have direct access to the archives I am about to roll out Asterisk to a few companies and would like to hear your experiences about the various handsets/phones that are Asterisk compatible I am primarily looking for 2 options, the first being a cheaper model which will provide reliability whilst still maintaining a reasonable feature set, and a reliable model from the more expensive range with more features But the definite focus here is on reliability and ease of maintenance Any help or advice would be greatly appreciated; I would really like to hear your experiences/recommendations Cheers Gary And middle posting is almost as bad. :-) But.. To the point... If you would have read what you were replying to, you would have noticed they did mention why weren't they 100Mbits connections on the 102 models for daisy chaining to a PC. Robert ___ 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] Reccomendation for reliable handsets
On February 14, 2005 09:23 am, Robert Webb wrote: On Mon, 14 Feb 2005 14:11:15 + Bob Goddard [EMAIL PROTECTED] wrote: On Monday 14 February 2005 13:00, Brett, Gary wrote: Thanks Mark I am definitely interested in the budgetone 102 but am a little concerned about the 10mbit only Ethernet ports !! From what I have read, these are relatively new models and I like the addition of a second port to daisy chain your PC from the same network connection, however why 10mbits and not 100mbits ??, I would have thought this would be a minimum these days, I don't know anyone who still runs 10mbits to the desktop, and im not too happy about bottlenecking my customers fast Ethernet network with these phones A real shame really. Does anybody know if Grandstream will be addressing this or indeed if they have any current models with at least 100mbit ports Please do not top post. I don't think there is a single IP phone which can flood a 10Mbps port. You do not need 100Mbps on a phone unless it has a passthrough to a PC. Let's see, using a 64Kbps codec and being generous, will use 100Kbps on the wire. Assuming the 10Mbps port can reliably run at 8Mbps, that means the phone would have to have 8 * 1000 / 100 = 80 concurrent RTP streams. Can anyone see anything wrong with my rough calculations? B -Original Message- From: Mark Benson [mailto:[EMAIL PROTECTED] Sent: 03 February 2005 14:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Reccomendation for reliable handsets I have been using an IN1002 generic handset (supposed to be an unbranded cisco copy but I am skeptical) for a few months (6months+) now, and it seems pretty stable - however I haven't found a reliable supplier Also there is almost no support for them.. I have switched to the grandstream budgetone 102 and they seem pretty good too. You can pretty much plug in and forget it with both phones. They do lock up occasionally (once a month to once every 3 months). I have yet to upgrade the firmware on the grandstreams... Mark Brett, Gary wrote: Sorry to move this up the list again, but does anybody have any advice on this -Original Message- From: Brett, Gary [mailto:[EMAIL PROTECTED] Sent: 02 February 2005 10:49 To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] Reccomendation for reliable handsets Hi there I'm sure this question has been raised a number of times before, but unfortunately I do not have direct access to the archives I am about to roll out Asterisk to a few companies and would like to hear your experiences about the various handsets/phones that are Asterisk compatible I am primarily looking for 2 options, the first being a cheaper model which will provide reliability whilst still maintaining a reasonable feature set, and a reliable model from the more expensive range with more features But the definite focus here is on reliability and ease of maintenance Any help or advice would be greatly appreciated; I would really like to hear your experiences/recommendations Cheers Gary And middle posting is almost as bad. :-) But.. To the point... If you would have read what you were replying to, you would have noticed they did mention why weren't they 100Mbits connections on the 102 models for daisy chaining to a PC. Robert ___ 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 Yes but failing to trim is even worse. :-) -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] Q: Does anyone have a WE multi-line card dialer phone working with *?
[EMAIL PROTECTED] wrote: Folks, I recently obtained a Western Electric multi-line phone and am seeking help with getting this beast working with *. The interesting stuff in my * implementation consists of a T100P card, a TDM400P card, and an Adtran TA750 channel bank with three quad-port FXS modules and a quad-port FXO. The TA750 is wired to a 24-port Cat 5 patch panel via a 25-pair Amp cable. The phone is a model 2662A1M; it has five lines, a hold button (I presume), card dialer capability, and a 25-pair Amp cable for connecting to The Phone System. (The card dialer feature, IMHO, scores major geek points. If you're not familiar with it, you take a special plastic card about the size of a credit card and punch out two tiny discs for each digit in a phone number. When it's time to call that number, you insert the card in the phone, take the handset off hook, push the START button, and--voila!--the phone speed dials your party.) Each line in the phone uses three pairs in the Amp cable; the first pair is for ring and tip, the second pair is a mystery (I'm eagerly awaiting a copy of one of the phone's BSPs so I can find out), and the third pair illuminates the lamp in the button. Most of the remaining pairs in the Amp cable connect to one of the terminal boards inside the phone, and one pair connects to the phone's network (presumably for common ringing, since the leads connect to L1 and L2). This, and MANY other multi line phones were used in 1A2 key systems You will need a support Key Service Unit to use this in a multi line with hold an illumination configuration Simply put your 2nd pair is the A lead control When the line is selected, and the phone is off hook, the pair is a short, allowing the A relay in the 400 ( NOT TDM400 ) card in the KSU to pick up, sending 10 VAC to the lamp leads. When the hold button is pressed, the TR loop is still made, but the A lead circuit is opened, putting the 400 card in the KSU into a hold condition, then when the hold button is released, the line button pops up as well, removing the set from the TR of that line. Many 1A2 multi line phones could be connected in parallel, in an office. The last 5 pair on this phone were probably used for outboard speakerphone, and would not be used or connected to other sets. Elementary Telephony. John Novack ___ 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] Linphone / Kphone
On Mon, 2005-02-14 at 22:29 +1100, Duane wrote: I gave up trying to use linux soft clients they all seem to have some fatal flaws or issues I could never fully get rid of While I'd second that, Gnomemeeting is still pretty good and by far the best softphone I've used on Linux. Currently, it supports H323 only, but SIP support is in development. It looks like it will take some more time though until a first test version is available. http://mail.gnome.org/archives/gnomemeeting-list/2004-December/msg00198.html ___ 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] Reccomendation for reliable handsets
And middle posting is almost as bad. :-) But.. To the point... If you would have read what you were replying to, you would have noticed they did mention why weren't they 100Mbits connections on the 102 models for daisy chaining to a PC. Robert SNIP Yes but failing to trim is even worse. :-) -A. Point well taken. ;-) Robert ___ 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 jitter?
Eric Wieling wrote: joachim wrote: Yes, It's untested and unfinished and touches the core of asterisk. (maybe causing massive amounts of deadlocks). So? That's what CVS-HEAD is there for. Adding in experimental patches willy-nilly, especially ones that have the potential to cause huge problems, confounds attempts to isolate bugs and test functionality. Mark does a pretty good job of keeping the HEAD version solid enough to use in production, as most of us running it on a daily basis can attest. What stops you from applying the patches to your own copy, and then playing with it to your heart's content--like the rest of us? It would work just like it had really been put into CVS-HEAD. less testers less bug reports for production use is stable version (asterisk doesnt have good roadmap and versioning :( ) --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ 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] Digium Cards connecting to BT
Brett, Gary wrote: Hi there Just a general question, has anybody experienced any problems with any Digium telephony cards in the UK, specifically with BT (British Telecom) lines. I just want to make sure there are no compatibility issues before purchasing cards, (mainly TDM400P's) Any comments would be greatly appreciated Can't comment on the UK, but you should be aware that the TDM400 is pretty insistent about PCI 2.2, and the FXS module does NOT provide a ground start configuration. Not a problem if you want just a POTS phone connected, but if you were to attempt to set up a two way trunk to a conventional PBX, you can only provide a loop start line. SOME versions of Asterisk had the loop pulse function not working as well. Perhaps none of this matters to you, but several collectors are attempting to use the Asterisk box as an interface to interconnect our old switches via the internet, and find the support for such basic telephony somewhat lacking. Interestingly enough, it seems the chipset used on the TDM 400 module supports a ground start configuration, but Digium chose not to make that available. John Novack ___ 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@home .5 and meetme
I'm having some problems getting meetme to work now that I have upgraded to .5 I am able to conference calls but every time I try to manage the conference through meetme it just says No users in this conference Any ideas why it doesn't see the conference call? Thanks for any help! Jason This message along with any attachments is intended only for the use of the individual or entity to which it was addressed. It may contain information that is confidential and prohibited from disclosure. If you are not the intended recipient, you are hereby notified that any dissemination or copying of this message or attachment is strictly prohibited. If you have received this message in error, please notify the original sender immediately by telephone or return e-mail and delete this message along with any attachments from this computer.. ___ 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] soho fax suggestions?
On Feb 13, 2005, at 4:43 PM, John Novack wrote: I use JFAX which I think is also known as Efax. If you are open to a new fax number anywhere else in the US from your home Zip code, then it is free. Otherwise there is a quarterly fee. AFAIK, you can't port an existing number to them, but I could be off on that. http://www.j2.com/jconnect/twa/page/servicesOverview I have a free eFax number that I've maintained for testing...although I'm unable to fax to it via Sixtel (you begin to hear a carrier but within 1/2 a second it's cut off). So much for testing. I have also used a Broadvox residential account for inbound faxing (they include fax-to-email as part of their feature set). But I think they may have broken this feature recently when they switched to a new VM system. While you might not be able to port a phone number to eFax, there's nothing stopping you from forwarding a number to eFax. But like I said, I've found outbound fax to be more of a problem than inbound. While the latter has worked well for me with Vonage and Voicepulse, the bigger problem is the former (outbound) as it's only ever worked reliably for me with a plain residential single-line account that I've had since May 2003. With Broadvox faxing was completely unreliable and often didn't work EVEN THOUGH they have T.38 support. Here's what I learned though: just because your CPE supports T.38 and your provider's gateway supports T.38, that doesn't mean that the carrier sitting in between supports T.38. Level 3, for instance, doesn't support T.38 at the moment (at least, not in all markets). So IMHO, T.38 ain't gonna do anyone any good until it's implemented across the board and who the heck knows when that might happen. While eFax, and similar services, are some sort of a solution to at least half the problem, I just think using these services is a kludge. The beauty of fax is: stick a document in at one end, dial a number, and the document spits out at the other end. No clumsy scanning and emailing involved. And while some folks think Fax is dying, I just don't agree. I think the technology needs to be rebuilt for IP, but I don't think the concept is going to go away anytime soon. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reccomendation for reliable handsets
I wouldn't recommend the grandstreams, I had very bad experience using the grandstream 102, It kep locking up on me. The buttons are very bad buttons. The sound quality is just as bad. ___ 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] soho fax suggestions?
On Feb 13, 2005, at 7:50 PM, Rich Adamson wrote: Can't offer any clue on the above either. Based on Steve Underwood's comments earlier (relative to outbound fax now fails on the TDM when it was working earlier), it would almost sound like a timing issue of some sort that is associated with calls initiated within *. Interesting. I wasn't aware of that. I'm more inclined to blame my CPE at the moment. Will probably switch to a Sipura 2100 soon. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable
I your case the problem is with the grandstream, the GS will not
display callerID correctly, take out the name from the callerid string
like this:
exten = ${EXTEN},PRI,SetCallerID(${CALLERIDNUM})
On Fri, 11 Feb 2005 23:46:13 -0800, Robert L Mathews
[EMAIL PROTECTED] wrote:
Nicol?s Gudi?o [EMAIL PROTECTED] wrote:
Paul, 1.0.5 stable suffers from caller id issues as well, at least for
SIP channels. What fixed things for me was swapping in app_dial.c from
1.0.2 stable (didn't try others). You could also just diff app_dial.c
between versions to find the problem but I took the lazy way out the
first time around.
Drumkilla reverted the callerid changes on the latest stable (thanks
Russell!). You will be fine if you checkout stable from CVS now.
Hmmm; I think I'm still having problems with it, using a completely
fresh checkout and compile:
Connected to Asterisk CVS-v1-0-02/11/05-17:34:08
I have two Zap FXS lines and two SIP phones, and:
- Zap channel to Zap channel, caller ID works (displays correctly on the
analog phone display).
- SIP phone to Zap channel, caller ID works.
- SIP phone to ZIP phone, caller ID does NOT work (Grandstream phone
displays Err).
- Zap channel to SIP phone, caller ID does NOT work.
- Incoming Free World Dialup calls to Zap channel extension, caller ID
works.
- Incoming Free World Dialup calls to SIP phone extension, caller ID
does NOT work.
So it seems that asterisk stable, as of today, does not send correct
caller ID on calls that end up on SIP phones, unless I'm doing something
boneheaded (although I used almost-identical config files on 1.0.2 with
no trouble).
A tcpdump shows that asterisk is sending this in the SIP INVITE header
to the phone:
From: asterisk sip:[EMAIL PROTECTED];
(IP address obscured; it's correct in the original.) But somehow
asterisk appears instead of the correct caller ID. Wasn't that the bug
other people were seeing that the stable update was supposed to fix?
Have I missed something obvious?
--
Robert L Mathews, Tiger Technologies http://www.tigertech.net/
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Re: [Asterisk-Users] ATA's
On Feb 14, 2005, at 5:39 AM, Nicolas Bougues wrote: On Sun, Feb 13, 2005 at 07:43:06PM -0600, Matthew Boehm wrote: We have had a big success with the Linksys PAP2-NA. 2 FX ports and 1 WAN port. Only downside is that only 1 call can be using 729 at a time. This has been confirmed with Linksys. They will be releasing PAP2-NAv2 in March to overcome this. In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. Personally, I dislike the lack of LEDs on the 2100. My 2100s have 3 LEDs, plus 2 for each RJ-45 port. Instead of just 2 for the SPA-2000. I think Matthew was referring to the lack of leds on the front of the Sipura. I can't seem to figure out why these manufacturers insist on building these boxes like you're going to stick them on your desk next to your phone. I want something that's more suitable for a phone closet. Too bad the PAP2-NA can't be purchased retail anymore. Then again, you're probably better off with a Sipura-branded unit anyhow. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 841 and paging function
nope, it uses an callinfo header: http://lists.digium.com/pipermail/asterisk-users/2005-January/086462.html On Mon, 14 Feb 2005 19:41:23 +0800, Craig Guy [EMAIL PROTECTED] wrote: I was browsing through the web config of a Sipura SPA-841 (Firmware 2.0.13) and noticed a setting marked 'paging' under supplementary services on the Phone settings page on the advanced admin login. Anyone know how it might be used? Could it be like the Snom - exten = 10,1,SetVar(VXML_URL=intercom=true) exten = 10,2,Dial(SIP/testuser) Craig ___ 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] Asterisk-H323
Greetings, I have a problem making a call from Asterisk to Cisco H323 PSTN gateway using H323 channel. I can call but there are no sound in both way. If I call H323 gateway directly from SJPhone I have no problem with sound. Any advice are welcome. Thanks in advance. ___ 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] FW: SER Asterisk Voicemail
Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. Original Message From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: FW: SER Asterisk Voicemail Date: Thu, 10 Feb 2005 16:45:53 - Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java Rockx using sipsak for sending mwi sip notify messages to the phone but is there a simpler way which I am blindly ignoring?? Thank you in advance, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ 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] E1-PRI: Warning Message: Unable to handle ROSE operation 36
hi, since my latest libpri update i get these messages: !! Unable to handle ROSE operation 36 !! Unable to handle ROSE operation 30 i searched through ITU X.219 and X.229 but can't find any values for the Remote Operations Service Elements. are these AOC-E messages? regards frank ___ 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] Digium Cards connecting to BT
Hi there Just a general question, has anybody experienced any problems with any Digium telephony cards in the UK, specifically with BT (British Telecom) lines. I just want to make sure there are no compatibility issues before purchasing cards, (mainly TDM400P's) Any comments would be greatly appreciated I know of about a dozen UK users, myself included, who cannot get the TDM400 FXO modules to do hangup detection correctly on a BT line. I have raised this with Digium support and they have suggested a fix, as has another user on this list. I haven't had time to test either fix yet, but will post to the list if I am successful. Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] speech recognition
daniela is affear but shhh - Original Message - From: Bill Maidment [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 14, 2005 8:54 AM Subject: Re: [Asterisk-Users] speech recognition David D. Faerman wrote: hi i am looking for some info for speech recognition for example when someone call to my house asterisk ask for who hi want to call and he say the name david or susan (wife) or daniela etc... And the wife asks Who's Daniela? ;-) -- _/_/_/_/ _/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/ _/ Bill Maidment Maidment Enterprises Pty Ltd Unless you are named Alfred E. Newman, you may read only the odd numbered words (every other word beginning with the first) of the message above. If you have violated that, then you hereby owe the sender AU$10 for each even numbered word you have read. Adapted from Stupid Email Disclaimers (see http://www.goldmark.org/jeff/stupid-disclaimers/) ___ 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] soho fax suggestions?
Mark Eissler wrote: While eFax, and similar services, are some sort of a solution to at least half the problem, I just think using these services is a kludge. I don't agree. Inbound faxes sent to my E-mail as TIFF are the best solution. No wasted paper, ink or toner. It it needs to be printed for some reason, the option is there, but forwarding to others is all done easily. If one views inbound and outbound as separate tasks, then this is perfect for inbound. No reason to link the two. Outbound can be handled a number of different ways, from scanning , to maintaining a standalone fax machine. JMO John Novack ___ 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] ATA that actually work with T.38
Hi, I am implementing T.38, and finding a problem getting boxes that work with T.38 for testing. A lot (maybe most) ATAs now claim to support T.38, but I'm finding a lot of these lie. I have one box here that just crashes when it hears a fax tone. :-) I'm looking for boxes known to implement T.38 properly, and which really work in the real world. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@home .5 and meetme
Hi Jason, The web meetme wont control a conference until someone dials in to it (eg you cant have a web interface setup then wait for someone to dial in afterwards). If you are unable to use the amp extension based conference rooms set up one of your own by editing the conf file and see if you can get web meetme to control this one (this is the way I use V 0.5 and it works fine). Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nash, Jason Sent: Monday, February 14, 2005 9:55 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] [EMAIL PROTECTED] .5 and meetme I'm having some problems getting meetme to work now that I have upgraded to .5 I am able to conference calls but every time I try to manage the conference through meetme it just says No users in this conference Any ideas why it doesn't see the conference call? Thanks for any help! Jason This message along with any attachments is intended only for the use of the individual or entity to which it was addressed. It may contain information that is confidential and prohibited from disclosure. If you are not the intended recipient, you are hereby notified that any dissemination or copying of this message or attachment is strictly prohibited. If you have received this message in error, please notify the original sender immediately by telephone or return e-mail and delete this message along with any attachments from this computer.. ___ 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] APP_QUEUE MYSQL LOGGING
Does anyone know if this has been implemented? I have been around the sites and haven't really found much. I know there was an old patch that would make it work but it doesn't do anything but break the application now. .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office ___ 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] Reccomendation for reliable handsets
Bob, Thanks for your reply, im not sure what top posting is, but I have been on holiday and am simply replying to a response that was given to my original question, If you could explain to me how I go about continuing the thread it would be much appreciated, with regards to your reply, I am indeed daisy chaining to the PC, hence my post point regarding bottlenecking the 100mbits to the desktop I just find it hard to understand the point of releasing a phone with a 2 port hub yet still limiting to ports to 10mbits, anyway , I take it there are no alternatives from budgetone, so I will have to at other low cost models Thanks all -Original Message- From: Bob Goddard [mailto:[EMAIL PROTECTED] Sent: 14 February 2005 14:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Reccomendation for reliable handsets On Monday 14 February 2005 13:00, Brett, Gary wrote: Thanks Mark I am definitely interested in the budgetone 102 but am a little concerned about the 10mbit only Ethernet ports !! From what I have read, these are relatively new models and I like the addition of a second port to daisy chain your PC from the same network connection, however why 10mbits and not 100mbits ??, I would have thought this would be a minimum these days, I don't know anyone who still runs 10mbits to the desktop, and im not too happy about bottlenecking my customers fast Ethernet network with these phones A real shame really. Does anybody know if Grandstream will be addressing this or indeed if they have any current models with at least 100mbit ports Please do not top post. I don't think there is a single IP phone which can flood a 10Mbps port. You do not need 100Mbps on a phone unless it has a passthrough to a PC. Let's see, using a 64Kbps codec and being generous, will use 100Kbps on the wire. Assuming the 10Mbps port can reliably run at 8Mbps, that means the phone would have to have 8 * 1000 / 100 = 80 concurrent RTP streams. Can anyone see anything wrong with my rough calculations? B -Original Message- From: Mark Benson [mailto:[EMAIL PROTECTED] Sent: 03 February 2005 14:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Reccomendation for reliable handsets I have been using an IN1002 generic handset (supposed to be an unbranded cisco copy but I am skeptical) for a few months (6months+) now, and it seems pretty stable - however I haven't found a reliable supplier Also there is almost no support for them.. I have switched to the grandstream budgetone 102 and they seem pretty good too. You can pretty much plug in and forget it with both phones. They do lock up occasionally (once a month to once every 3 months). I have yet to upgrade the firmware on the grandstreams... Mark Brett, Gary wrote: Sorry to move this up the list again, but does anybody have any advice on this -Original Message- From: Brett, Gary [mailto:[EMAIL PROTECTED] Sent: 02 February 2005 10:49 To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] Reccomendation for reliable handsets Hi there I'm sure this question has been raised a number of times before, but unfortunately I do not have direct access to the archives I am about to roll out Asterisk to a few companies and would like to hear your experiences about the various handsets/phones that are Asterisk compatible I am primarily looking for 2 options, the first being a cheaper model which will provide reliability whilst still maintaining a reasonable feature set, and a reliable model from the more expensive range with more features But the definite focus here is on reliability and ease of maintenance Any help or advice would be greatly appreciated; I would really like to hear your experiences/recommendations Cheers Gary ___ 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] Call parking
You have to add the include statement in the context thet you want the parking (park, and pickup) to be available. # will only work with a t (for the called), and/or a T (for the caller) in the dial command. On Sun, 13 Feb 2005 00:28:30 -0500, Robert Webb [EMAIL PROTECTED] wrote: I am trying to figure out call parking. It is my understanding that it is built into *. I have edited the features.conf like I want it but am unsure where to add the include statement. Right now if I am on a call from the FXO bridged to the FXS port and I hit the # key, nothing happens. I have tried reading the wiki but cannot find anything that clearly explains this feature. Robert ___ 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] Asterisk as SIP UAC !!!
Hi gentleman I've configured SER to forward every call starting with sip uri request 1 to Asterisk. I need to configure Asterisk as a Sip UAC in order to make it call to my other SIP Provider outside my network, sending username and password for authentication. I've read at www.voip-info.org some articles but found none that could suit to my needs, but yet I've found an article which explains an implementation very similiar to what I need (http://www.voip-info.org/wiki-Asterisk+voicepulse+connect), but in my solution, I don't use IAX just sip terminatino via Internet. I've tried to do exactly as this tutorial said, but with one difference, all the entries at iax.conf I've made at sip.conf. The result is that I can still connect my sip phone to my server but it doesn't give me an outside line after I press 1. Have anyone implemented this solution or know what I may be doing wrong ?? My configurations are following below: Extensions.conf exten = 1,1,Dial(SIP/username:password@go2call,30,rT) exten = 2,1,Playback(tt-weasels) exten = 2,2,Hangup() exten = 3,1,Playback(tt-weasels) Sip.conf [go2call] context = go2call username=username secret=password auth=md5 type=friend host=go2callhost -- Felipe Martins TEP Solution New Technologies Mundivox Communications [EMAIL PROTECTED] Site: www.mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 ___ 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] ztmonitor
Good day list, I am feeling extra stupid this Monday morning and am hoping someone can come to the rescue. I am trying to use the ztmonitor utility on my wildfire 4 FXO card. and have read the following from the wiki. *Wiki start If you set this to yes, use ztmonitor to adjust the rxgain and txgain. Ztmonitor isn't installed by default; but it is included with the Zaptel source code, so look in /usr/src/zaptel. Use ztmonitor like this: ./ztmonitor 1 -v Rx ##Tx Place a call, let the remote party talk, then adjust them until they sound better. For example, try this setting in zapata.conf: rxgain=10.5 txgain=-4.5 Note: If you set the txgain value too low, your outbound calls may not go thorugh since the DTMF tones are too quiet to be picked up. ***Wiki End** if I use txgain=0; rxgain=-; this I get the following: Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ##*#* Then I Edit /etc/asterisk/Zapata.conf if I use txgain=0; rxgain=7.5; this I get the following: Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ##*#* I get basically the same thing Can anyone tell me if I am missing something Oh yeah , I tell asterisk to restart after editing the Zapata.conf file. Thanks ron oledata.mso Description: Binary data oledata.mso Description: Binary data ___ 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] ztmonitor SOLVED
Sorry issue solved. I had to RTFM better I just needed to increase the gain higher my magic number ended up being 15.5 Sorry to bug 8000 ppl. ~ron -Original Message- From: Ronald Hartmann [mailto:[EMAIL PROTECTED] Sent: Monday, February 14, 2005 11:18 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztmonitor Good day list, I am feeling extra stupid this Monday morning and am hoping someone can come to the rescue. I am trying to use the ztmonitor utility on my wildfire 4 FXO card. and have read the following from the wiki. *Wiki start If you set this to yes, use ztmonitor to adjust the rxgain and txgain. Ztmonitor isn't installed by default; but it is included with the Zaptel source code, so look in /usr/src/zaptel. Use ztmonitor like this: ./ztmonitor 1 -v Rx ##Tx Place a call, let the remote party talk, then adjust them until they sound better. For example, try this setting in zapata.conf: rxgain=10.5 txgain=-4.5 Note: If you set the txgain value too low, your outbound calls may not go thorugh since the DTMF tones are too quiet to be picked up. ***Wiki End** if I use txgain=0; rxgain=-; this I get the following: Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ##*#* Then I Edit /etc/asterisk/Zapata.conf if I use txgain=0; rxgain=7.5; this I get the following: Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ##*#* I get basically the same thing Can anyone tell me if I am missing something Oh yeah , I tell asterisk to restart after editing the Zapata.conf file. Thanks ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk-H323
Cisco and Asterisk are not behind firewall. Where can I check for settings noH245Tuneling and noFastStart in Asterisk H323? - -- Executing Dial(SIP/msn-069a, H323/[EMAIL PROTECTED]:1720) in new stack -- Called [EMAIL PROTECTED]:1720 -- H323/peer:1720 is making progress passing it to SIP/msn-069a -- H323/peer:1720 is ringing -- H323/peer:1720 answered SIP/msn-069a == Spawn extension (messanger, 73952389506, 1) exited non-zero on 'SIP/msn-069a' -- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, February 14, 2005 11:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk-H323 Make sure settings for: noH245Tuneling and noFastStart parameters are correctly tuned both sides. Is Cisco or Asterisk behind NAT? Send more info Greetings, I have a problem making a call from Asterisk to Cisco H323 PSTN gateway using H323 channel. I can call but there are no sound in both way. If I call H323 gateway directly from SJPhone I have no problem with sound. Any advice are welcome. Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk-H323
Hi there, The settings are in oh323.conf ; Enable fast start (yes,no). ; fastStart=yes ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=yes ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; I assume you are using the OH323 driver right? Also if no audio, it could also be a codec issue. You need to set the codec for the OH323 call in oh323.conf as well. David Hong Kong On Mon, 14 Feb 2005 11:27:53 -0500, Vitalie Apostu wrote Cisco and Asterisk are not behind firewall. Where can I check for settings noH245Tuneling and noFastStart in Asterisk H323? - -- Executing Dial(SIP/msn-069a, H323/[EMAIL PROTECTED]:1720) in new stack-- Called [EMAIL PROTECTED]:1720-- H323/peer:1720 is making progress passing it to SIP/msn-069a-- H323/peer:1720 is ringing -- H323/peer:1720 answered SIP/msn-069a == Spawn extension (messanger, 73952389506, 1) exited non-zero on 'SIP/msn-069a' -- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, February 14, 2005 11:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk-H323 Make sure settings for: noH245Tuneling and noFastStart parameters are correctly tuned both sides. Is Cisco or Asterisk behind NAT? Send more info ___ 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] ATA that actually work with T.38
Hi Gary, Aren't those all tied to service providers now? Regards, Steve Gary Carr wrote: We use the PAP-2NA with fax machines and have not had any problems. Gary Hi, I am implementing T.38, and finding a problem getting boxes that work with T.38 for testing. A lot (maybe most) ATAs now claim to support T.38, but I'm finding a lot of these lie. I have one box here that just crashes when it hears a fax tone. :-) I'm looking for boxes known to implement T.38 properly, and which really work in the real world. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Cards connecting to BT
The X101P works but I dont think it would be acceptable in a commercial environment. The audio levels are too low and there is too much echo (or speech break-up with the aggressive cancellation set on). Saying that hang-up detection works and CLID works with some source code changes. Anybody got a winning setup in the UK, I'd love to hear from you if so! ? Mike On Mon, 14 Feb 2005 15:13:37 -, Patrick Lidstone (Personal E-mail) [EMAIL PROTECTED] wrote: Hi there Just a general question, has anybody experienced any problems with any Digium telephony cards in the UK, specifically with BT (British Telecom) lines. I just want to make sure there are no compatibility issues before purchasing cards, (mainly TDM400P's) Any comments would be greatly appreciated I know of about a dozen UK users, myself included, who cannot get the TDM400 FXO modules to do hangup detection correctly on a BT line. I have raised this with Digium support and they have suggested a fix, as has another user on this list. I haven't had time to test either fix yet, but will post to the list if I am successful. Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA that actually work with T.38
No, the PAP2's are. The PAP2-NA is for any provider. Gary - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 14, 2005 11:33 AM Subject: Re: [Asterisk-Users] ATA that actually work with T.38 Hi Gary, Aren't those all tied to service providers now? Regards, Steve Gary Carr wrote: We use the PAP-2NA with fax machines and have not had any problems. Gary Hi, I am implementing T.38, and finding a problem getting boxes that work with T.38 for testing. A lot (maybe most) ATAs now claim to support T.38, but I'm finding a lot of these lie. I have one box here that just crashes when it hears a fax tone. :-) I'm looking for boxes known to implement T.38 properly, and which really work in the real world. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Uptime/reliability with SER, Asterisk
Could anyone shed any light on how SER and/or Asterisk (stable branch) has held up for them in that last while? Are you using SER and/or * in a production environment? Do you ever restart the software or reboot the system? How many users are utilizing the system? How many calls per day/concurrently? I read some uptimes and such on the mailing list from long ago, so I was wondering what some more recent results were like. I'm running Asterisk at home, but only since recently so my experience won't be a good representation of the reliability and stability. Thanks in advance. ___ 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] Broadvoice international dialling question
On Sun, 13 Feb 2005, Malcolm Taylor wrote:
I'd be grateful if someone could point me in the right direction.
I have a Broadvoice trunk attached to Asterisk which I use for frequent
calls to the UK using the following in extensions.conf
exten = _0[1-68].,1,Ringing
exten = _0[1-68].,2,Dial(SIP/BV/01144${EXTEN:1})
exten = _0[1-68].,3,Hangup
The caller hears immediate ringing, though it seems that Broadvoice takes a
long time to make the international connection and sometimes fails
altogether
This is because you've told Asterisk to play a ringing sound before it has
even attempted to place the call with BV. Take out your Ringing line and
that behavior should stop.
Greg
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RE: [Asterisk-Users] Asterisk-H323
noH245Tunneling instead of noH245Tuneling
typedef struct call_options {
charcid_num[80];
charcid_name[80];
int noFastStart;
int noH245Tunneling;
int noSilenceSuppression;
unsigned intport;
int progress_setup;
int progress_alert;
int progress_audio;
int dtmfcodec;
} call_options_t;
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Monday, February 14, 2005 11:29 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk-H323
Make sure settings for:
noH245Tuneling and noFastStart parameters are correctly tuned both sides.
Is Cisco or Asterisk behind NAT?
Send more info
Greetings,
I have a problem making a call from Asterisk to Cisco H323 PSTN
gateway using H323 channel. I can call but there are no sound in both
way. If I call
H323 gateway directly from SJPhone I have no problem with sound.
Any advice are welcome.
Thanks in advance.
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Re: [Asterisk-Users] ATA that actually work with T.38
Quoting Gary Carr [EMAIL PROTECTED]: You might want to tell that to these guys: http://www.voipsupply.com/product_info.php?products_id=317 regards, Paul No, the PAP2's are. The PAP2-NA is for any provider. Gary - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 14, 2005 11:33 AM Subject: Re: [Asterisk-Users] ATA that actually work with T.38 Hi Gary, Aren't those all tied to service providers now? Regards, Steve Gary Carr wrote: We use the PAP-2NA with fax machines and have not had any problems. Gary Hi, I am implementing T.38, and finding a problem getting boxes that work with T.38 for testing. A lot (maybe most) ATAs now claim to support T.38, but I'm finding a lot of these lie. I have one box here that just crashes when it hears a fax tone. :-) I'm looking for boxes known to implement T.38 properly, and which really work in the real world. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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 -- [EMAIL PROTECTED] http://www.fielding.ca - This mail sent through IMP: http://horde.org/imp/ ___ 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] soho fax suggestions?
Maxemail.com is out there too. $14.95/yr if you don't care about the number, or $6/month if you do. Not a bad deal for the service. Outbound is still the most difficult, but there are print-fax drivers out there. Packetel has (or used to have) a $4/month option as well, iirc -Original Message- From: Mark Eissler [mailto:[EMAIL PROTECTED] Sent: Monday, February 14, 2005 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] soho fax suggestions? On Feb 13, 2005, at 4:43 PM, John Novack wrote: I use JFAX which I think is also known as Efax. If you are open to a new fax number anywhere else in the US from your home Zip code, then it is free. Otherwise there is a quarterly fee. AFAIK, you can't port an existing number to them, but I could be off on that. http://www.j2.com/jconnect/twa/page/servicesOverview I have a free eFax number that I've maintained for testing...although I'm unable to fax to it via Sixtel (you begin to hear a carrier but within 1/2 a second it's cut off). So much for testing. I have also used a Broadvox residential account for inbound faxing (they include fax-to-email as part of their feature set). But I think they may have broken this feature recently when they switched to a new VM system. While you might not be able to port a phone number to eFax, there's nothing stopping you from forwarding a number to eFax. But like I said, I've found outbound fax to be more of a problem than inbound. While the latter has worked well for me with Vonage and Voicepulse, the bigger problem is the former (outbound) as it's only ever worked reliably for me with a plain residential single-line account that I've had since May 2003. With Broadvox faxing was completely unreliable and often didn't work EVEN THOUGH they have T.38 support. Here's what I learned though: just because your CPE supports T.38 and your provider's gateway supports T.38, that doesn't mean that the carrier sitting in between supports T.38. Level 3, for instance, doesn't support T.38 at the moment (at least, not in all markets). So IMHO, T.38 ain't gonna do anyone any good until it's implemented across the board and who the heck knows when that might happen. While eFax, and similar services, are some sort of a solution to at least half the problem, I just think using these services is a kludge. The beauty of fax is: stick a document in at one end, dial a number, and the document spits out at the other end. No clumsy scanning and emailing involved. And while some folks think Fax is dying, I just don't agree. I think the technology needs to be rebuilt for IP, but I don't think the concept is going to go away anytime soon. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-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] Italian speaking. Asterisk configuration and needs
Hi, is there someone who speaks in Italian? I'll try to explain in english my problem, but if there is someone who speaks italian i think it would be better for me. I'd like to use asterisk only as IVR and call diverting. I have only one phone line, and no other phones, all the calls arrive at one number. I would like something that answare, and depending from the 'street' followed by the ivr, it diverts the call to an other phone number. What do I need for implementing it? I have a server whith Linux suse, an ADSL connection, a telephone line. Thanks to everyone that can help me, and also to the others :-) __ Tiscali Adsl 3 Mega Flat, 3 MESI GRATIS! Con Tiscali Adsl 3 Mega Flat navighi in Rete alla supervelocita' a soli 29.95 euro al mese senza limiti di tempo. Attivati entro il 15 Febbraio 2005, 3 MESI sono GRATIS Scopri come http://abbonati.tiscali.it/adsl/sa/2flat_tc/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk-H323
No, I am using H323 driver -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Liu Sent: Monday, February 14, 2005 11:36 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk-H323 Hi there, The settings are in oh323.conf ; Enable fast start (yes,no). ; fastStart=yes ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=yes ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; I assume you are using the OH323 driver right? Also if no audio, it could also be a codec issue. You need to set the codec for the OH323 call in oh323.conf as well. David Hong Kong On Mon, 14 Feb 2005 11:27:53 -0500, Vitalie Apostu wrote Cisco and Asterisk are not behind firewall. Where can I check for settings noH245Tuneling and noFastStart in Asterisk H323? - -- Executing Dial(SIP/msn-069a, H323/[EMAIL PROTECTED]:1720) in new stack-- Called [EMAIL PROTECTED]:1720-- H323/peer:1720 is making progress passing it to SIP/msn-069a-- H323/peer:1720 is ringing -- H323/peer:1720 answered SIP/msn-069a == Spawn extension (messanger, 73952389506, 1) exited non-zero on 'SIP/msn-069a' -- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, February 14, 2005 11:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk-H323 Make sure settings for: noH245Tuneling and noFastStart parameters are correctly tuned both sides. Is Cisco or Asterisk behind NAT? Send more info ___ 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] Linphone / Kphone / lipz4
On Mon, 2005-02-14 at 13:08 +0100, Jens Kübler wrote: Maybe you wanna check out the softphone zip4x5 made by Zultys. It's the software which is used by the same hardphone. Howdy, Do you use this product and do you have any relationship with Zultys? It looks interesting, but it is documented to support only old RedHat versions and they don't release source to let me recompile. I am not a big RedHat fan, but if I have to use it on the desktop, I would want something newer than RedHat 9. If you can tell me you are using it with a newer distro, that would help. Have a good day, Ralph ___ 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] connect asterisk to ISDN in China
Dear Xu, my name is Marco Castillo, I'm in Guatemala, Central America, and I have recently succesfully installed a TE110P here in Guatemala. There are many implementations of a E1 or T1, but I think that the great majority can be configured via the zaptel drivers. I will suggest you to buy a card and make the leap of faith!!! Regards Marco -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Xu, Duo Sent: Sunday, February 13, 2005 12:58 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] connect asterisk to ISDN in China Hi, I plan to install asterisk and connect it to telco through ISDN in China. I'd love to know if the ISDN standard in China has any difference than in America before I buy the digium card. anybody has experience in it? or anybody who installed asterisk with ISDN in asia can share their expierience? Or, can anybody give me some links to educate me ISDN knowledge about the difference in China? (My heard there is something different there, but i dont know the details.) Thanks __ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] E1-PRI: Warning Message: Unable to handle ROSE operation 36
On Mon, 14 Feb 2005, Frank Sautter wrote: since my latest libpri update i get these messages: !! Unable to handle ROSE operation 36 !! Unable to handle ROSE operation 30 i searched through ITU X.219 and X.229 but can't find any values for the Remote Operations Service Elements. are these AOC-E messages? The AOC-elements are enumerated in Q.956 clause 2.7.2. From what I can tell they are AOC-E in Charging Units and a ChargingRequest (perhaps a charging request reply). Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 registration
Hi all, How can I configured H323 EPs or OH323 EPs to get them authenticated through GNUGK??? Many thanks Ben ___ 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] Who makes these phones?
I have 3 of the Black ones. I think the are junk. They work, and I actually found a manual online for it. I ran into a weird problem last week. After I did a Reset to Factory. All the phones were getting th same IP address from the DHCP server, I found that the MAC address on the phones were the same, I was able to MANUALLY set them to what ever I wanted, and go them to work. Kyle ___ 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] Linphone / Kphone
Darren Ellis wrote: Hi, I have * working with X-Lite and Sipura adapters, but I have one person who is linux based, and is trying to use Linphone and Kphone. His end works, but I get very bad echo on my end. Have any of you folks been able to get linux based soft phones working well with *? I'd appreciate links to howtos/docs if you have them, and/or samples of working configs for * and the linux softphones. Thanks much for sharing your experience. Darren ___ 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 I used to use kphone and have very bad echo, I switched to sjphone and it worked great. Kyle ___ 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] (no subject)
I have a question for using gastman. I have set up extensions for my IAX users as IAX2/username, and I keep getting the following Dunno how to tell if IAX2/username/6 is IAX2/username I was wondering if there is some sort of wildcard character that can be used here? The number changes every time, so I do not think that I can put in seperate extensions. Thank You, Ron Frederick ___ 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] ATA that actually work with T.38
That site is correct. You have to be authorized by Linksys to order the product from a distributor but they will work with any VoIP service. We use them with our * service. Gary Quoting Gary Carr [EMAIL PROTECTED]: You might want to tell that to these guys: http://www.voipsupply.com/product_info.php?products_id=317 regards, Paul No, the PAP2's are. The PAP2-NA is for any provider. Gary - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 14, 2005 11:33 AM Subject: Re: [Asterisk-Users] ATA that actually work with T.38 Hi Gary, Aren't those all tied to service providers now? Regards, Steve Gary Carr wrote: We use the PAP-2NA with fax machines and have not had any problems. Gary Hi, I am implementing T.38, and finding a problem getting boxes that work with T.38 for testing. A lot (maybe most) ATAs now claim to support T.38, but I'm finding a lot of these lie. I have one box here that just crashes when it hears a fax tone. :-) I'm looking for boxes known to implement T.38 properly, and which really work in the real world. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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 -- [EMAIL PROTECTED] http://www.fielding.ca - This mail sent through IMP: http://horde.org/imp/ ___ 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] Who makes these phones?
Hi! http://www.broadbandphone.com.au/global/pnp.htm they are called a Kitty Ethernet Phone, seem to be available in 3 or 4 models but with identical Guts. The only info I have found on them is Gateway Technologies, supposedly the Chinese manufacturer website... http://www.ipgw.net/EN/index.htm The phone on the left (BBP GW01) is also known as Giptel G100, Siptronic ST-100, Yuxin YWH10 A(b) or # ViDa i Phone-D00. It can also be run with IAX2 firmware. Read more: http://www.voip-info.org/wiki-Giptel+IP+phones http://www.voip-info.org/wiki-PA168 Cheers, Philipp ___ 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: [Serusers] FW: SER Asterisk Voicemail
If the message is only sent as an email attachment (delete=yes,attach=yes) then the user must listen to it by playing the attached wav file on their pc. If the message is saved on the Asterisk server then you need to provide dial-in access to Asterisk that sends the caller to VoiceMailMain. From there they can access their mailbox and manage messages. _Steve Aisling O'Driscoll wrote: Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. Original Message From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: FW: SER Asterisk Voicemail Date: Thu, 10 Feb 2005 16:45:53 - Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java Rockx using sipsak for sending mwi sip notify messages to the phone but is there a simpler way which I am blindly ignoring?? Thank you in advance, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[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] Sipura g729 call quality to PSTN
If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side (the audio on the phone connected to the Sipura sounds fine). My guess is that the Sipura does not compress the outbound audio very effectively and since the incoming audio from the PSTN is already compressed by the VoIP provider, it is just delivering the good-sounding g729 stream. It is worth noting that call quality on both the IP and PSTN side is great when using the Cisco 7960 with g729. It is just with the Sipura that the sound quality on the PSTN-side sounds like a bad quality cell phone call. I even got an SPA-2100 in hopes that the g729 would sound better on that unit, but the same issue is present there as well. Is it just a bad implementation of g729 compression with the Sipura product line? Any thoughts or recommendations are appreciated :) Thanks! - Pedro ___ 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] Uptime/reliability with SER, Asterisk
Our SER/Asterisk implementation is extremely stable if you define stable as the ability to deliver a set of features without either application crashing. We are a production environment with 75 users total. Asterisk is only used for voicemail. The only issue we have is that the audio (greeting or message) being play from Asterisk sometimes has a robotic or stuttering quality to it. I suspect this is latency in the data network but I have yet to figure it out. -Steve Dana Olson wrote: Could anyone shed any light on how SER and/or Asterisk (stable branch) has held up for them in that last while? Are you using SER and/or * in a production environment? Do you ever restart the software or reboot the system? How many users are utilizing the system? How many calls per day/concurrently? I read some uptimes and such on the mailing list from long ago, so I was wondering what some more recent results were like. I'm running Asterisk at home, but only since recently so my experience won't be a good representation of the reliability and stability. Thanks in advance. ___ 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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[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] cdr_mysql losing logs
I noticed a problem this morning with our cdr logging. We have a cron
job that places a call file into the spool directory having asterisk
call itself to check to make sure its still handling incoming calls
correctly, then queries the CDR database in mysql and makes sure that
appropriate records exist.
I can confirm that the call is happening correctly, but I'm missing
records in the database:
select calldate, disposition, lastapp, channel from cdr where clid =
xx order by calldate desc limit 45;
| 2005-02-14 11:34:04 | ANSWERED| Hangup | Zap/2-1 |
| 2005-02-14 11:34:03 | ANSWERED| BackGround | Zap/3-1 |
| 2005-02-14 11:32:03 | ANSWERED| Hangup | Zap/1-1 |
| 2005-02-14 11:30:04 | ANSWERED| Hangup | Zap/3-1 |
| 2005-02-14 11:30:02 | ANSWERED| BackGround | Zap/4-1 |
Notice the missing BackGround entry from the 11:32 call.
The asterisk console logs for this same duration:
^M-- Attempting call on Zap/g1/2144680768 for [EMAIL PROTECTED]:1 (Retry 1)
Using channel 1
Urgent handler
Urgent handler
^M-- Remote UNIX connection disconnected
^M-- Accepting call from '' to '2144680768' on channel 0/3, span 1
Enabled echo cancellation on channel 3
Launching 'Goto'
^M-- Executing Goto(Zap/3-1, neospire|s|1) in new stack
^M-- Goto (neospire,s,1)
Launching 'Wait'
^M-- Executing Wait(Zap/3-1, 1) in new stack
Difference is 1120, ms is 160
Write returned -1 (Resource temporarily unavailable) on channel 2
Write returned -1 (Resource temporarily unavailable) on channel 2
Write returned -1 (Resource temporarily unavailable) on channel 2
Write returned -1 (Resource temporarily unavailable) on channel 2
Launching 'Answer'
^M-- Executing Answer(Zap/3-1, ) in new stack
Urgent handler
Launching 'Wait'
^M-- Executing Wait(Zap/3-1, 1) in new stack
Enabled echo cancellation on channel 1
Dropping duplicate answer!
^MChannel Zap/1-1 was answered.
Launching 'StopMonitor'
^M-- Executing StopMonitor(Zap/1-1, ) in new stack
Launching 'Answer'
^M-- Executing Answer(Zap/1-1, ) in new stack
Launching 'Playback'
^M-- Executing Playback(Zap/1-1, 30seconds) in new stack
Set channel Zap/1-1 to write format gsm
Scheduling timer at 160 sample intervals
^M-- Playing '30seconds' (language 'en')
Launching 'DigitTimeout'
^M-- Executing DigitTimeout(Zap/3-1, 5) in new stack
^M-- Set Digit Timeout to 5
Launching 'ResponseTimeout'
^M-- Executing ResponseTimeout(Zap/3-1, 10) in new stack
^M-- Set Response Timeout to 10
Launching 'BackGround'
^M-- Executing BackGround(Zap/3-1, neo-welcome-options) in new stack
Set channel Zap/3-1 to write format gsm
Scheduling timer at 160 sample intervals
^M-- Playing 'neo-welcome-options' (language 'en')
Scheduling timer at 0 sample intervals
Scheduling timer at 0 sample intervals
Set channel Zap/3-1 to write format ulaw
Scheduling timer at 0 sample intervals
Scheduling timer at 0 sample intervals
Set channel Zap/1-1 to write format ulaw
Launching 'Hangup'
^M-- Executing Hangup(Zap/1-1, ) in new stack
Spawn extension (neospire,6501,4) exited non-zero on 'Zap/1-1'
cdr_mysql: inserting a CDR record.
cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcont
ext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,ac
countcode,uniqueid,userfield) VALUES ('2005-02-14
11:32:03','2147201442','214720
1442','6501','neospire',
'Zap/1-1','','Hangup','',16,16,'ANSWERED',3,'','1108402
321.10557','')
Hanging up channel 'Zap/1-1'
zt_hangup(Zap/1-1)
Set option AUDIO MODE, value: ON(1) on Zap/1-1
Hangup: channel: 1 index = 0, normal = 13, callwait = -1, thirdcall = -1
Not yet hungup... Calling hangup once with icause, and clearing call
Urgent handler
disabled echo cancellation on channel 1
Set option TDD MODE, value: OFF(0) on Zap/1-1
Updated conferencing on 1, with 0 conference users
Set option AUDIO MODE, value: OFF(0) on Zap/1-1
disabled echo cancellation on channel 1
^M-- Hungup 'Zap/1-1'
Urgent handler
Call completed to Zap/g1/2144680768
^M-- Channel 0/3, span 1 got hangup
cdr_mysql: inserting a CDR record.
cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcont
ext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,ac
countcode,uniqueid,userfield) VALUES ('2005-02-14
11:32:01','','','s','neospire'
,
'Zap/3-1','','BackGround','neo-welcome-options',18,17,'ANSWERED',3,'','1108402
321.10558','')
Hanging up channel 'Zap/3-1'
zt_hangup(Zap/3-1)
Set option AUDIO MODE, value: ON(1) on Zap/3-1
Hangup: channel: 3 index = 0, normal = 15, callwait = -1, thirdcall = -1
Not yet hungup... Calling hangup once with icause, and clearing call
Urgent handler
disabled echo cancellation on channel 3
Set option TDD MODE, value: OFF(0) on Zap/3-1
Updated conferencing on 3, with 0 conference users
Set option AUDIO MODE, value: OFF(0) on Zap/3-1
disabled echo cancellation on channel 3
^M-- Hungup 'Zap/3-1'
Notice that
Re: [Asterisk-Users] Uptime/reliability with SER, Asterisk
I really appreciate your reply. For Asterisk, are you using G729 as your codec, or something more high-bandwidth (ulaw)? Is there any definition of stable that you would use that would point to SER and Asterisk not being stable? Again, thanks for your reply. -- Dana On Mon, 14 Feb 2005 13:27:53 -0500, Steve Blair [EMAIL PROTECTED] wrote: Our SER/Asterisk implementation is extremely stable if you define stable as the ability to deliver a set of features without either application crashing. We are a production environment with 75 users total. Asterisk is only used for voicemail. The only issue we have is that the audio (greeting or message) being play from Asterisk sometimes has a robotic or stuttering quality to it. I suspect this is latency in the data network but I have yet to figure it out. -Steve Dana Olson wrote: Could anyone shed any light on how SER and/or Asterisk (stable branch) has held up for them in that last while? Are you using SER and/or * in a production environment? Do you ever restart the software or reboot the system? How many users are utilizing the system? How many calls per day/concurrently? I read some uptimes and such on the mailing list from long ago, so I was wondering what some more recent results were like. I'm running Asterisk at home, but only since recently so my experience won't be a good representation of the reliability and stability. Thanks in advance. ___ 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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[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] H323 no sound
Could you help me with this problem? When I call H323 gateway there is no
sound in both ways.
Here is h323 debug:
- begin
-- Executing Dial(SIP/msn-6297, H323/[EMAIL PROTECTED]:1720) in new
stack
Allowed Codecs:
Table:
G.729A{sw} 1
G.729{sw} 2
G.711-uLaw-64k 3
G.711-ALaw-64k 4
UserInput/hookflash 5
UserInput/RFC2833 6
Set:
0:
0:
G.729A{sw} 1
G.729{sw} 2
G.711-uLaw-64k 3
G.711-ALaw-64k 4
1:
UserInput/hookflash 5
2:
UserInput/RFC2833 6
-- Making call to [EMAIL PROTECTED]:1720 without gatekeeper.
== New H.323 Connection created.
-- root is calling host [EMAIL PROTECTED]:1720
--Call token is ip$localhost/31515
-- Call reference is 31515
-- DTMF Payload is 101
-- Called [EMAIL PROTECTED]:1720
-- Sending SETUP message
-- Transmitting RFC2833 on payload 101
-- Started logical channel: sending G.729A{sw}
-- channelsOpen = 1
External RTP Session Starting
RTP channel id 1 parameters:
-- remoteIpAddress: 81.17.12.22
-- remotePort: 26454
-- ExternalIpAddress: 0.0.0.0
-- ExternalPort: 14182
-- Started logical channel: receiving G.729A{sw}
-- channelsOpen = 2
External RTP Session Starting
RTP channel id 1 parameters:
-- remoteIpAddress: 81.17.12.22
-- remotePort: 26454
-- ExternalIpAddress: 0.0.0.0
-- ExternalPort: 14182
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
=-= In OnAlerting for call 31515: sessionId=0
-- Ringing phone for 73952389512
- Progress Indicator: 8
-- H323/peer:1720 is making progress passing it to SIP/msn-6297
-- H323/peer:1720 is ringing
-- Transmitting RFC2833 on payload 101
=-= In OnConnectionEstablished for call 31515
-- Connection Established with Unknown
-- H323/peer:1720 answered SIP/msn-6297
-- Received Facility message...
-- Received Facility message...
-- Received Facility message...
-- Received Facility message...
-- ClearCall: Request to clear call with token ip$localhost/31515,
cause 3
-- Sending RELEASE COMPLETE
channelsOpen = 1
channelsOpen = 0
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
== Spawn extension (messanger, 73952389512, 1) exited non-zero on
'SIP/msn-6297'
-- ClearCall: Request to clear call with token ip$localhost/31515,
cause 7
-- Unknown has cleared the call
== H.323 Connection deleted.
end
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Re: [Asterisk-Users] Digium Cards connecting to BT
On Mon, 14 Feb 2005 15:13:37 -, Patrick Lidstone (Personal E-mail) wrote: Hi there Just a general question, has anybody experienced any problems with any Digium telephony cards in the UK, specifically with BT (British Telecom) lines. I just want to make sure there are no compatibility issues before purchasing cards, (mainly TDM400P's) Any comments would be greatly appreciated I know of about a dozen UK users, myself included, who cannot get the TDM400 FXO modules to do hangup detection correctly on a BT line. I have raised this with Digium support and they have suggested a fix, as has another user on this list. I haven't had time to test either fix yet, but will post to the list if I am successful. Patrick I have intermittent problems as well. Patrick - if you do find a solution please share it with the rest of us. If it doesn't work then perhaps it might be an idea if those of us with problems penned a joint email to Digium it might get a more concerted response. I'm not suggesting that Digium isn't being unhelpful or ignoring the problem, but if enough people don't voice their concerns then it will not get prioritised for a fix. Regards, George ___ 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] Broadvoice international dialling question
Many thanks Greg!
Sometimes things are just too obvious!
Malcolm
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill
Sent: Monday, February 14, 2005 11:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice international dialling question
On Sun, 13 Feb 2005, Malcolm Taylor wrote:
I'd be grateful if someone could point me in the right direction.
I have a Broadvoice trunk attached to Asterisk which I use for frequent
calls to the UK using the following in extensions.conf
exten = _0[1-68].,1,Ringing
exten = _0[1-68].,2,Dial(SIP/BV/01144${EXTEN:1})
exten = _0[1-68].,3,Hangup
The caller hears immediate ringing, though it seems that Broadvoice takes
a
long time to make the international connection and sometimes fails
altogether
This is because you've told Asterisk to play a ringing sound before it has
even attempted to place the call with BV. Take out your Ringing line and
that behavior should stop.
Greg
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Re: [Asterisk-Users] Linphone / Kphone
On Mon, 2005-02-14 at 10:47 -0700, Kyle Hagan wrote: I used to use kphone and have very bad echo, I switched to sjphone and it worked great. It isn't too bad, but it has latency (compare it e.g. to asterisk as softphone and you'll see what I mean) and no dial pad. So I found it isn't really satisfying either. Another point to note is that seemingly all closed source softphones (SJ, XLite beta and also cornfed) make connections to web servers and transmit platform/call information. Don't know how you think about that, but for me that's behavior I'd like to avoid if ever possible. Regards, Bruno. ___ 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@Home ... the next step
So I've got it installed and running (?) except for one error message and I haven't had time research it yet but I'd like to get a quick reply or pointer to my next step to getting [EMAIL PROTECTED] working. The error is during boot ( Linux ) and comes from ztcfg ( I think? Memory going quickly ) about FXS and FXO being configured the opposite of what one thinks. Do I need to have a line connected/plugged-in to the TDM400P to get it to configure? And is there a specific _next_ place ( URL/URL/Wiki ) to continue to get [EMAIL PROTECTED] configured? Actually I'm testing at home since it is not considered a good thing to experiment with our business' lines. :-) TIA, Rod -- --- [This E-mail scanned for viruses by Declude Virus] ___ 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] Linphone / Kphone
On Mon, 14 Feb 2005 20:01:18 +0100, Bruno Hertz [EMAIL PROTECTED] wrote: Another point to note is that seemingly all closed source softphones (SJ, XLite beta and also cornfed) make connections to web servers and transmit platform/call information. Don't know how you think about that, but for me that's behavior I'd like to avoid if ever possible. Regards, Bruno. Do you have this documented somewhere? Is this for the Linux Xlite and SJphone only, or the Win32 ones as well? -- Dana ___ 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] Bristuff-0.2.0-RC5 florz patched weird error and no outgoing calls?
I applied the florz patch but my problems remain. Now I get all sorts of weird errors on the console and I cannot make outgoing calls. Incoming calls work as expected. I am using a single HFC-S card with BRI. Any clue what these errors below are? Ri = 44651 TEI msg = 3 TEI = 7f Ri = 3800 TEI msg = 3 TEI = 7f Ri = 42399 TEI msg = 3 TEI = 7f Ri = 42409 TEI msg = 3 TEI = 7f Ri = 22078 TEI msg = 3 TEI = 7f Ri = 991 TEI msg = 3 TEI = 7f Feb 14 10:30:25 NOTICE[14777]: app_dial.c:762 dial_exec: Unable to create channel of type 'Zap' Ri = 36942 TEI msg = 3 TEI = 7f Feb 14 10:31:13 WARNING[14777]: pbx.c:444 pbx_exec: Stack overflow, cannot create another stack Ri = 25084 TEI msg = 3 TEI = 7f Feb 14 10:34:21 WARNING[14777]: chan_zap.c:7411 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 received TEI check request for TEI = 102 Feb 14 10:39:01 WARNING[14777]: chan_zap.c:7411 zt_pri_error: PRI: !! Not good - head of queue has not been transmitted yet Feb 14 10:39:33 WARNING[14777]: chan_zap.c:7411 zt_pri_error: PRI: !! Got reject for frame 8, but we have nothing -- resetting! ___ 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] Intermediary jitter buffering
Yea, I might be doing native bridging. The peer might do jitter buffering (as its Asterisk), or they might have it turned off for whatever reason. Also, my clients have significantly more jitter issues (Guatemala ISPs suck), so it's possible that I might want a different jitterbuffer setup than my provider usually does. At any rate, I (as a VoIP service provider) want be able to have the most amount of control over quality settings like this. -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kann Sent: Sunday, February 13, 2005 8:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Intermediary jitter buffering On Feb 12, 2005, at 9:10 PM, Michael Giagnocavo wrote: Hello, I understand that only the destination of a call should do jitter buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no transfers), PhoneA and PhoneB need to perform their own jitter buffering, and Asterisk will just forward the frames, correct? What happens if the peer does not support jitter buffering, but is close by so there's no need for jitter buffering? My situation is that I have an Asterisk machine right in front of our provider's systems (same switch, 1ms latency). If they don't have jitter buffering, how can I force my Asterisk machine to jitter buffer calls from my users to them? Assuming this is all IAX, presently, the jitterbuffer is either on, or off, as you configure; it doesn't go off automatically if it's in the middle of a bridge (although native bridging does bypass it). So, in your situation, with the current code, disable native bridging, and enable the jitterbuffer, and you should get it. But, we're working on improving this area a lot; this is an uncommon situation, though: Why doesn't the peer support jitterbuffering? -SteveK ___ 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] Outgoing analog problems and questions with quicknet cards
I've been fighting this for a while and have come back to the list with
some of my configuration information. I have a quicknet internet linejack
card and have been thus far unsuccessful at placing outbound calls over
the analog phone line. I can receive calls through the line jack and
route them through sip phones but can't get out through the linejack.
I'm under the impression that you can only do one or the other with one of
these cards, however I've yet to accomplish dialing out on an analog line.
My phone.conf is below
[interfaces]
mode=immediate
;mode=dialtone
;mode=fxo
mode=fxs
context=default
format=slinear
txgain=100%
rxgain=1.0
device = /dev/phone0
from what I've read, fxo is for receiving while fxs is for sending analog.
From my testing, changing this doesn't seem to make a difference in the
phone.conf
My extensions.conf file is below. I'm definately confused on what
LOCALTRUNK should be defined as. Phone calls coming in over the linejack
card are on the channel Phone/phone0. I'm assuming the same is true if
I'm using that card for dialing out instead.
[globals]
LOCALTRUNK=Phone/phone0
[hayden]
exten = s,1,Dial(SIP/paul,20,tr)
exten = s,2,NoOp(${CALLERID})
include = outgoing
[outgoing]
ignorepat = 9
exten = _XX,1,Dial(${LOCALTRUNK}/${EXTEN})
I've read a lot of the documentation and have tried to focus more on
channel configuration today. I'm confused about the zapata and
zaptel.conf files. I'm under the impression that they're for using the
zaptel cards and not the linejack cards. Do I need to have either of
these files?
I've been at this for quite a while, and it's safe to say I'm pretty stuck
with my current hardware setup. Any insight is much appreciated as I'd
love to be able to pick up my work phone from the house and make calls
through it from my house as well.
Hayden Myers
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