Re: [Asterisk-Users] some EICON Diva 4BRI questions
On Fri, April 21, 2006 15:19, Armin Schindler said: Hi Klaus, Thanks for the detailed answers and isdn for Linux basics. I will take the opportunity to ask some more questions :-) On Fri, April 21, 2006 12:24, Armin Schindler said: On Fri, 21 Apr 2006, Klaus Darilion wrote: 3. Is the PIN layout for TE or NT mode? It is TE PIN layout, you need a crossed cable with 100Ohm termination for NT-mode. Why do I need a cable with 100Ohm termination? Shouldn't be the termination inside the DIVA Server? Until now (with quadbri and other isdn card) I only used CAT5 cables with BRI-crossover PIN layout. No resistors. Can you please explain this a little bit more or give me links to the wiring basics? I must admit that I don't really know that. Maybe the quadbri has this termination automatically on board. Since the Eicon DIVA Server card has basically a TE port, the termination is necessary. I always use the 100Ohm termination in my NT-cross-cables. Here is a short description on Melware Wiki: http://www.melware.org/BriCrossCable Hi Armin! I guess the RJ-45 jack with the resistor must be connected to the DIVA card, right? But I'm still confused. Usually, if a line needs termination, the termination is needed on both ends. Thus, if there is no line termination inside the DIVA card, we would termination in TE mode too. regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Separating Asterisk SIP extensions from dialing each other.
There is a way to have same extensions on the same system. Just put them in seperate contexts within extensions.conf and assign the appropriate extensions in sip.conf, iax.conf,... to these. Alex On Fri Apr 21 16:41 , Rick Smith [EMAIL PROTECTED] sent: This is coming from an * noob. :) I've got two customers, they both are replacing their phone systems with VOIP, and we need to retain both their existing dialplans. One has 5 extensions starting at 100, and the other has 10 extensions, starting at 100. Is there a way to have the same extension number twice in the same asterisk system ? They will have different incoming DIDs of course. I don't want them to be able to see / hear / feel / dial each other internally, either. They must remain completely independent. If anyone's got pointers in a Wiki or PDF somewhere, let me know. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations
On 4/21/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi listers, I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's. I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office. However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price. I would appreciate any input people have to offer. James, 24 ports PoE on a Linksys switch occupied with Polycom IP 501's, no problem. The Polycom's use approx. 4W, so the 7.5W per port (if you use more then 12) is no problem at all. Agreed, a more expensive switch (Cisco,HP,Alcatel) might be better, but the Linksys is worth the money! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting to a cluster of SIP servers
My Asterisk server is connecting to sip.plus.net, which resolves to multiple IP addresses: sip.plus.net. 300 IN A 84.92.0.75 sip.plus.net. 300 IN A 84.92.0.76 sip.plus.net. 300 IN A 84.92.5.189 sip.plus.net. 300 IN A 84.92.5.190 If one of these machines is down (i.e. it's not replying to the SIP packets or it's sending back ICMP Port Unreachable), Asterisk keeps trying the same server. Shouldn't Asterisk move on to the next server automatically in this case? It seems to only way to do this at the moment is to run the reload command, which causes it to do a DNS lookup and it may then pick one of the other servers. -- - Steve xmpp:[EMAIL PROTECTED] sip:[EMAIL PROTECTED] http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.2.7.1 on FC5 won't make install
I think that you are speaking about the sound-addon right? Did you use the "make install" option on the sounds? Alex On Sat Apr 22 4:26 , Cliff Savage [EMAIL PROTECTED] sent: The make seems to go okay. [[EMAIL PROTECTED] asterisk-1.2.7.1]# uname -a Linux somebox.org 2.6.16-1.2080_FC5smp #1 SMP i686 i686 i386 GNU/Linux mkdir -p /var/lib/asterisk/sounds/digits mkdir -p /var/lib/asterisk/sounds/priv-callerintros for x in sounds/digits/*.gsm; do \ if grep -q "^%`basename $x`%" sounds.txt; then \ install -m 644 $x /var/lib/asterisk/sounds/digits ; \ else \ echo "No description for $x"; \ exit 1; \ fi; \ done No description for sounds/digits/*.gsm make: *** [datafiles] Error 1 All the folders in /var/lib/asterisk/sounds are full of sounds except digits and priv-caller-intros. Those 2 folders are empty. Manually copy them in and recompile maybe??? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
Hello, I am about to put an asterisk server between the telco E1 and our old Matra PBX. Should I use an ethernet cross cable? Something else? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: SPA 3000 - UK Replacement
First off I am totally annoyed and let down by PC World Business (PCWB part of the Dixons Group). I ordered one of these babies from them over a month ago. After constantly chasing them up they finally told me they couldn't deliver, and have now only just returned the money they stole from me. I only bought from them because they showed a 4-day availability stock level! Now I'm screwed as it seems these are impossible to come by in the UK now since Sipura decided to discontinue it.. Now my back to my subject. Does anyone know of a decent replacement for the SPA3000. I need at least 1 FXO and 1 FXS but am willing to pay for 2 FXS on the same unit. What I'm looking for needs to be of a likable price to the SPA3000 which in it's hayday was retailing for around £70 at some outlets. I'm primarily looking for something network attachable. But could stretch to USB or PCI if the price was right.. I'm steering away from PCI cards as they seem to have terrible issues with UK analogue lines such as not being able to detect hang ups.. (Also; the server I'd ideally like to add this capability too, has no free PCI slots..) Thanks for your time, Steve Daniels (I've been trying to send this for ages but crappy outlook keep sending it from another of my email accounts despite the fact that I keep setting it to send it via the correct one!) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: None To: [EMAIL PROTECTED] Subject: Microsoft Office Outlook Test Message This is an e-mail message sent automatically by Microsoft Office Outlook's Account Manager while testing the settings for your POP3 account. attachment: winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting to a cluster of SIP servers
Although there maybe a better way, this would work: 1. Add the IP's into your sip.conf and set qualify=yes. 2. Make your dialplan something like the following: exten = _X.,1,Dial,SIP/[EMAIL PROTECTED] exten = _X.,2,Hangup exten = _X.,102,Dial,SIP/[EMAIL PROTECTED] exten = _X.,103,Hangup exten = _X.,203,Dial,SIP/[EMAIL PROTECTED] exten = _X.,204,Hangup exten = _X.,304,Dial,SIP/[EMAIL PROTECTED] exten = _X.,305,Hangup This would make your failover work but certainly wouldn't help with the load balancing between the servers. If any cannot qualify or are congested, they will automatically failover to the next server. I believe most people use an SER proxy for this type of application. It seems to work well with the round robin type DNS. William -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hill Sent: Saturday, April 22, 2006 5:13 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Connecting to a cluster of SIP servers My Asterisk server is connecting to sip.plus.net, which resolves to multiple IP addresses: sip.plus.net. 300 IN A 84.92.0.75 sip.plus.net. 300 IN A 84.92.0.76 sip.plus.net. 300 IN A 84.92.5.189 sip.plus.net. 300 IN A 84.92.5.190 If one of these machines is down (i.e. it's not replying to the SIP packets or it's sending back ICMP Port Unreachable), Asterisk keeps trying the same server. Shouldn't Asterisk move on to the next server automatically in this case? It seems to only way to do this at the moment is to run the reload command, which causes it to do a DNS lookup and it may then pick one of the other servers. -- - Steve xmpp:[EMAIL PROTECTED] sip:[EMAIL PROTECTED] http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: SPA 3000 - UK Replacement
Steven wrote: First off I am totally annoyed and let down by PC World Business (PCWB part of the Dixons Group). I ordered one of these babies from them over a month ago. After constantly chasing them up they finally told me they couldn't deliver, and have now only just returned the money they stole from me. I only bought from them because they showed a 4-day availability stock level! You think that's bad, I ordered one on the 10th of march from redstore, that was showing a 3-5 day. They still haven't despatched the unit and I have been trying to call them now (on their 0870 number) for about a week, during the past 3 weeks I have been sending them email after email that hasn't been responded to. Now I'm screwed as it seems these are impossible to come by in the UK now since Sipura decided to discontinue it.. Oh Balls! Didn't know that. Broadbandbuyer have them lsited as entering stock on the 25th of May, I'd order one if I knew that I could cancel the redstore one. Now my back to my subject. Does anyone know of a decent replacement for the SPA3000. I need at least 1 FXO and 1 FXS but am willing to pay for 2 FXS on the same unit. What I'm looking for needs to be of a likable price to the SPA3000 which in it's hayday was retailing for around £70 at some outlets. £49.45 + VAT at broadbandbuyer.co.uk or 78 euros from someone on ebay in the netherlands. I'm primarily looking for something network attachable. But could stretch to USB or PCI if the price was right.. I'm steering away from PCI cards as they seem to have terrible issues with UK analogue lines such as not being able to detect hang ups.. (Also; the server I'd ideally like to add this capability too, has no free PCI slots..) I'd be interested in a similar unit myself, if the price is right although I'd prefer a network attachable device myself. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SP3000 question
Hi, all I finally got myself one of those SIPURA boxes. It is labeled as Linksys, but this is actually a SP3000 box. Anyway, unit has lots of configuration parameters. Not all are obvious. At the moment it registers against my *, but all the calls I do from analog phone connected to it, go to VoIP channel. As this part is still in testing, I want all the outgoing calls got to PSTN by default and dial, say 0, to get an outside VoIP line. I would like to do it as part of SP3000 configuration, not as part of * dialplan. Can someone help me? Thanks, Rudolf P.S. I am reading the manual now, but it is rather large. I want to have SP3000 installed now, but to avoid screams from my wife, I want analog phone to work as an analog phone for now while I am playing with configuration and * dialplans. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: SPA 3000 - UK Replacement
Here is Grandstream's version of the spa-3000. I have used it and it works great with asterisk. http://grandstream.com/y-ht488.htm -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of tom Sent: Saturday, April 22, 2006 8:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: SPA 3000 - UK Replacement Steven wrote: First off I am totally annoyed and let down by PC World Business (PCWB part of the Dixons Group). I ordered one of these babies from them over a month ago. After constantly chasing them up they finally told me they couldn't deliver, and have now only just returned the money they stole from me. I only bought from them because they showed a 4-day availability stock level! You think that's bad, I ordered one on the 10th of march from redstore, that was showing a 3-5 day. They still haven't despatched the unit and I have been trying to call them now (on their 0870 number) for about a week, during the past 3 weeks I have been sending them email after email that hasn't been responded to. Now I'm screwed as it seems these are impossible to come by in the UK now since Sipura decided to discontinue it.. Oh Balls! Didn't know that. Broadbandbuyer have them lsited as entering stock on the 25th of May, I'd order one if I knew that I could cancel the redstore one. Now my back to my subject. Does anyone know of a decent replacement for the SPA3000. I need at least 1 FXO and 1 FXS but am willing to pay for 2 FXS on the same unit. What I'm looking for needs to be of a likable price to the SPA3000 which in it's hayday was retailing for around £70 at some outlets. £49.45 + VAT at broadbandbuyer.co.uk or 78 euros from someone on ebay in the netherlands. I'm primarily looking for something network attachable. But could stretch to USB or PCI if the price was right.. I'm steering away from PCI cards as they seem to have terrible issues with UK analogue lines such as not being able to detect hang ups.. (Also; the server I'd ideally like to add this capability too, has no free PCI slots..) I'd be interested in a similar unit myself, if the price is right although I'd prefer a network attachable device myself. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] confused about iax and voip providers termination
On Friday 21 April 2006 16:29, T. Shaw wrote: After getting an account and following their steps, I can make calls out using my IAX (cubix) and Sip (Xlite) phones. However, I'm a bit confused on the purpose on how my box asterisk box is involved. I completely turned off my Asterisk box, and made a call out using either of my softphones and I was successful. So I gathered that the entire point of iax termination is solely for INBOUND calls TO ME (such if I have a DID). Otherwise I'm just using them as a proxy to forward my sip traffic to them directly from my desktop. If you configure your phone to use your provider directly you do not need Asterisk at all. Termination means that the provider is terminating calls you place through them. In effect, you send calls to them so they can originate them on the PSTN. Origination providers send calls to you that they are terminating for the PSTN. That's not the best description, no, but essentially it comes down to this: you send calls to a termination provider, and an origination provider sends calls to you. Asterisk is a PBX. It can route calls, handle voicemail, do IVR, make your coffee and mow your lawn if you're willing to put the time and equipment to the job. If all you want to do is send all of your calls out to ONE provider and not worry about anything else, you don't necessarily need Asterisk at all. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] anybody get experience with dell powerconnect 3424 and QOS for asterisk traffic?
Hi, Anyone already configured a powerconnect 34xx to prioritize voip traffic over the lan ? I just bought a 3424 for the lan and I like to give priority to voice , thanks in advance, jean-louis ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT.
hum, the last time i downloaded something every file has different dates. However, im looking at a new version that i have downloaded today: http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre9/libmfcr2-0.0.3.tar.gz And checking the source it seems that tar is the most recent version. I check the version looking in the C code for a fix i know must be there, in mfcr2.c line 2780, after the generation tone it must OR the signal with 0x80. Let me tell you that I have not tested that version. I have a custom version that i fixed (because it gave me the same error you have) and I sent the fix to Steve Underwood, but he told me that my fix was not error proof, and that may fail (I have 1 month now in a production server with no problems tough), so he made a similar fix, and told me that was more reliable. The link I just gave you is for the TAR with Steve Underwood fix. I guess you already contacted me off-list to quote you for my consultory. If you still have problems let me know and i may be able to help you through SSH. Best Regards On 4/21/06, Anton Krall [EMAIL PROTECTED] wrote: Moises, how can I find out which version Im running, on Steves ftp all say 0.0.3 or the date also says the same date. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Moises Silva |Sent: Friday, April 21, 2006 9:43 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT. | |A couple of weeks ago, libmfcr2 has a small error in the tone |signaling for the call setup, that was fixed 2 weeks ago or |so, please, wich version of libmfcr2 are you using? if you |dont know try upgrading to the latest version. Im pretty much |sure that you have the very same problem we had. | |Regards | |On 4/21/06, Jefferson Carvalho [EMAIL PROTECTED] wrote: | Hello All, | | I'm facing problems with Unicall on this scenario : | | CentOS 4.3 - Running on x86_64 | Asterisk 1.2.7.1 | Zaptel 1.2.5 | | When running zttool , shows all Spans OK. | | But I can't receive and make calls. | | I tried to change many parameters and still doesn't work. | | Any clues ? | | * unicall.conf | | [channels] | | language=br | | context=incoming-pstn | usecallerid=yes | hidecallerid=no | immediate=no | callwaitingcallerid=yes | threewaycalling=yes | transfer=yes | cancellforward=yes | callreturn=yes | echocancel=yes | echocancelwhenbridged=yes | | rxgain=0.0 | txgain=0.0 | faxdetect=both | loglevel=255 | protocolclass=mfcr2 | protocolvariant=br,20,4 | protocolend=cpe | group=1 | callgroup=1 | | channel = 1-15 | channel = 17-31 | channel = 32-46 | channel = 48-62 | channel = 63-77 | channel = 94-108 | channel = 110-124 | | * zaptel.conf * | | loadzone=br | defaultzone=br | | | span=1,1,0,cas,hdb3 | cas=1-15:1101 | cas=17-31:1101 | | span=2,0,0,cas,hdb3 | cas=32-46:1101 | cas=48-62:1101 | | | span=3,0,0,cas,hdb3 | cas=63-77:1101 | cas=79-93:1101 | | span=4,0,0,cas,hdb3 | cas=94-108:1101 | cas=110-124:1101 | | | | * lor error * | | -- Executing Dial(SIP/1000-1de2, Unicall/g1/40020022|40|Ttr) | in new stack Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 | unicall_report: MFC/R2 | UniCall/1 Call control(1) | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: | MFC/R2 | UniCall/1 Make call | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: | MFC/R2 | UniCall/1 Making a new call with CRN 32769 Apr 20 19:13:57 | WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 | UniCall/1 0001 - [1/ 1/Idle /Idle ] | -- Called g1/40020022 | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:2644 handle_uc_event: | Unicall/1 event Dialing | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 |unicall_report: MFC/R2 | UniCall/1 - [1/ 40/Seize /Idle ] | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 |unicall_report: MFC/R2 | UniCall/1 4 on - [2/ 40/Group I /Idle ] | Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 |unicall_report: MFC/R2 | UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS | ] cause | 32769 - T1 timed out | Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 |unicall_report: MFC/R2 | UniCall/1 4 off - [1/ 1/Idle /Idle ] | Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 |unicall_report: MFC/R2 | UniCall/1 1001 - [1/ 1/Idle /Idle ] | Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:2644 handle_uc_event: | Unicall/1 event Protocol failure | -- Unicall/1 protocol error. Cause 32769 Apr 20 19:14:02 | WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 | UniCall/1 Channel echo cancel | Apr 20 19:14:03 WARNING[30676]: chan_unicall.c:627 unicall_report: | MFC/R2 | UniCall/1 Channel gains | Apr 20 19:14:03
[Asterisk-Users] What about NCS and Asterisk?
Hi List!!I've been reading about NCS and Asterisk which I think it's very dificult to put them together. Is there any chances to registrer a Motorola SBV5120 , which work with NCS (variant of MGCP), with Asterisk? I know exists patches but they're really don't work for what I read on the mailing list. I think it could be possible to do that union between 5120 and * due to Asterisk supports MGCP and it's no so different from NCS. This is an old topic but without a complete solution.Thanks for any helpCarlos Bernat ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
A T carrier cable is not the same as an ethernet cable. A T carrier cable uses a real metal shielded RJ-45 and loosely twisted pair wire. With most modern T carrier equipment, you can use a CAT-5 ethernet cable instead of a real T carrier cable. A T-carrier crossover cable does not have the same wiring pattern as a crossover ethernet cable. With an older piece of equipment like the Matra, I would be tempted to purchase a real T carrier crossover cable. This is covered in my book, by the way. Louis-David Mitterrand wrote: Hello, I am about to put an asterisk server between the telco E1 and our old Matra PBX. Should I use an ethernet cross cable? Something else? Paul [EMAIL PROTECTED] www.signate.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: what cable to connect a legacy PBX to a TE410P ?
On Sat, Apr 22, 2006 at 08:09:13AM -0700, Paul Mahler wrote: A T carrier cable is not the same as an ethernet cable. A T carrier cable uses a real metal shielded RJ-45 and loosely twisted pair wire. With most modern T carrier equipment, you can use a CAT-5 ethernet cable instead of a real T carrier cable. A T-carrier crossover cable does not have the same wiring pattern as a crossover ethernet cable. With an older piece of equipment like the Matra, I would be tempted to purchase a real T carrier crossover cable. This is covered in my book, by the way. I'm just back from the client's site and failed to perform any tests. When connecting with a straight cat5 cable to the telco's E1 socket I got a red status led and RED alarm on the TE410. However when reconnecting the PBX's cable, the telco's socket led went back to green. On the other hand, I tried connecting the PBX's cable to the TE410 and got a green led and REC status. What could be wrong? My /etc/zaptel.conf: span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 span=3,1,0,ccs,hdb3,crc4 span=4,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone=fr defaultzone=fr My /etc/asterisk/zapata.conf: ;; to telco context=default signalling=pri_cpe group = 1 channel = 1-15 channel = 17-31 ;; to old pbx context=international signalling=pri_net group = 2 channel = 32-46 channel = 48-62 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] anybody get experience with dell powerconnect 3424 and QOS for asterisk traffic?
On 4/22/06, Jean-Louis curty [EMAIL PROTECTED] wrote: Hi,Anyone already configured a powerconnect 34xx to prioritize voip traffic over the lan ?I just bought a 3424 for the lan and I like to give priority to voice ,thanks in advance, jean-louis Hi, do you have another layer 3 QOS switch on premise? The 3424 is QOS aware, but does not support QOS tagging and prioritization. In essence, it will just honor and pass the tags. However, depending on what you have going on in your network, I doubt you will need QOS if you just have one switch. MHO of course. -Brian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
LOL -Original Message- From: Paul Mahler [mailto:[EMAIL PROTECTED] Sent: Sat 4/22/2006 11:09 AM To: asterisk-users@lists.digium.com Cc: Subject:Re: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ? A T carrier cable is not the same as an ethernet cable. A T carrier cable uses a real metal shielded RJ-45 and loosely twisted pair wire. With most modern T carrier equipment, you can use a CAT-5 ethernet cable instead of a real T carrier cable. A T-carrier crossover cable does not have the same wiring pattern as a crossover ethernet cable. With an older piece of equipment like the Matra, I would be tempted to purchase a real T carrier crossover cable. This is covered in my book, by the way. Louis-David Mitterrand wrote: Hello, I am about to put an asterisk server between the telco E1 and our old Matra PBX. Should I use an ethernet cross cable? Something else? Paul [EMAIL PROTECTED] www.signate.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
Can't anyone stop self-promotion and tell the poor guy what he needs. A T1/E1 X-over cable using an RJ-45 (8-cond.) is pinned out as follows: 1 - 4 2 - 5 3 - NU 4 - 1 5 - 2 6 - NU 7 - NU 8 - NU NU = Not Used I have not in my experience seen any problems with using a Good Quality Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you should be fine. As far as the shielding goes, I use UTP cables and Connectors all the time and some of my X-connects run over 100 feet. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, April 22, 2006 11:49 AM To: Paul Mahler; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ? LOL -Original Message- From: Paul Mahler [mailto:[EMAIL PROTECTED] Sent: Sat 4/22/2006 11:09 AM To: asterisk-users@lists.digium.com Cc: Subject:Re: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ? A T carrier cable is not the same as an ethernet cable. A T carrier cable uses a real metal shielded RJ-45 and loosely twisted pair wire. With most modern T carrier equipment, you can use a CAT-5 ethernet cable instead of a real T carrier cable. A T-carrier crossover cable does not have the same wiring pattern as a crossover ethernet cable. With an older piece of equipment like the Matra, I would be tempted to purchase a real T carrier crossover cable. This is covered in my book, by the way. Louis-David Mitterrand wrote: Hello, I am about to put an asterisk server between the telco E1 and our old Matra PBX. Should I use an ethernet cross cable? Something else? Paul [EMAIL PROTECTED] www.signate.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
Alexander Lopez wrote: Can't anyone stop self-promotion and tell the poor guy what he needs. Seems to me that SOME self promotion belongs on the biz list, and for those considered in the inner circle it is OK here! Everyone is equal. Some are more equal than others JMO John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?
I for one think it is a great idea to buy a book with hard earned money and wait a few days to a week just to get an answer to a question that is freely available on the internet immediately. Hard pressed to think of anything in the book that is not on with wiki with more up to date and useful information. -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED] Sent: Sat 4/22/2006 11:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ? Can't anyone stop self-promotion and tell the poor guy what he needs. A T1/E1 X-over cable using an RJ-45 (8-cond.) is pinned out as follows: 1 - 4 2 - 5 3 - NU 4 - 1 5 - 2 6 - NU 7 - NU 8 - NU NU = Not Used I have not in my experience seen any problems with using a Good Quality Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you should be fine. As far as the shielding goes, I use UTP cables and Connectors all the time and some of my X-connects run over 100 feet. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, April 22, 2006 11:49 AM To: Paul Mahler; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ? LOL -Original Message- From: Paul Mahler [mailto:[EMAIL PROTECTED] Sent: Sat 4/22/2006 11:09 AM To: asterisk-users@lists.digium.com Cc: Subject:Re: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ? A T carrier cable is not the same as an ethernet cable. A T carrier cable uses a real metal shielded RJ-45 and loosely twisted pair wire. With most modern T carrier equipment, you can use a CAT-5 ethernet cable instead of a real T carrier cable. A T-carrier crossover cable does not have the same wiring pattern as a crossover ethernet cable. With an older piece of equipment like the Matra, I would be tempted to purchase a real T carrier crossover cable. This is covered in my book, by the way. Louis-David Mitterrand wrote: Hello, I am about to put an asterisk server between the telco E1 and our old Matra PBX. Should I use an ethernet cross cable? Something else? Paul [EMAIL PROTECTED] www.signate.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3
I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone IncompatibleWith Asterisk 1.2.3
I am not familiar with that phone. Is it single pair? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sat 4/22/2006 12:13 PM To: asterisk-users@lists.digium.com Cc: Subject:[Asterisk-Users] PANASONIC KX-TS208W - Speakerphone IncompatibleWith Asterisk 1.2.3 I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2. winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?
I tend to use the soft sell. Looks like I Just self-promoted :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Saturday, April 22, 2006 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ? Alexander Lopez wrote: Can't anyone stop self-promotion and tell the poor guy what he needs. Seems to me that SOME self promotion belongs on the biz list, and for those considered in the inner circle it is OK here! Everyone is equal. Some are more equal than others JMO John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] server choice
I like IBM best but out of your choices, I would select the HP issam wrote: hello I will buy a server to make an IVR solution with asterisk and a te110p T1/E1 digium card. I have two options: 1/ HP Proliant ML370 G4 : Xeon 64bits 3,2Ghz, 1Go Ram, 3 disks SCSI 73Go 2/ Dell PowerEdge 2800: Xeon 64bits 3Ghz, 1Go Ram, 3 disks SCSI 73Go I use linux fedora core 3 and I want a help to choose a good server to use with asterisk Thank You ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone IncompatibleWith Asterisk 1.2.3
You got me there, it's at a customer's premise. I will have to find out from them, if it a single pair. -- Original message -- From: "Steve Totaro" [EMAIL PROTECTED] I am not familiar with that phone. Is it single pair?-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sat 4/22/2006 12:13 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone IncompatibleWith Asterisk 1.2.3 I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2. ---BeginMessage--- winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?
I agree. I haven't had a problem using CAT-5, even for long runs, however it's not a real T-Carrier cable and I didn't know how old the PBX is. Paul I have not in my experience seen any problems with using a Good Quality Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you should be fine. As far as the shielding goes, I use UTP cables and Connectors all the time and some of my X-connects run over 100 feet. Paul Mahler - [EMAIL PROTECTED] www.signate.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Orative
Never heard of them but the concept seems pretty good. I didnt look at the pricing but I asume that they will a chunk of thier money on the calling side (i.e. calling both parties and connecting them etc.). Another chunk on software and set up. I think such a solution is workable for asterisk however you would have to have good programing skills, lots of time. It seems the best way to go about it would be to run asterisk real time along with an external application that works with asterisk that keeps track of all users, thier status etc. So in the long run it is possible but it may take a little time. Also you would have to think of what you can offer that this company already dosent. (you can include this as an option when you build some one an asterisk system but is it worth all the time an effort etc. ?) Dean Collins [EMAIL PROTECTED] wrote:Has anyone heard anything about these guys? Anyone seen anything like this?http://www.orative.com/solutions.phpIts seems very cool, basically uses GPRS as a digital overlay on your mobile phone for additional functionality such as presence and IM though Im sure they have some other functionality (voicemail access, call announce etc) coming down the pipeline.Any thoughts, how hard would it be to build something like this from scratch for the asterisk platform?Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger with Voice. PC-to-Phone calls for ridiculously low rates.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
Alexander Lopez wrote: I have not in my experience seen any problems with using a Good Quality Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you should be fine. As far as the shielding goes, I use UTP cables and Connectors all the time and some of my X-connects run over 100 feet I have used cat-5 for everything communications. serial printers, dumb terminals, DS1 and even 10/100 ethernet. :-) It's easier to have it installed as a network jack and then use for whatever you need. ... Andrew McRory LinuxSystems Tallahasse, FL ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?
att(formerly SBC, formerly Southwestern Bell, formerly ATT) just came out and installed my PRI. FYI, they used Cat 5e cable. No special T1 cabling that costs a fortune to buy somewhere, just plain old Cat 5e cable. Guess what guys? If they are using this as customers' sites, they are probably using it elsewhere.It's only as good as the weakest link, so you can go out and spend lots of money on T1 cable, or just use Cat 5e like the telco guys do. -- Lacy MooreAspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3
[EMAIL PROTECTED] wrote: I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2. Replace the batteries! Alkaline only, replace every 6 months 1.2.3 is also defective for other reasons. Upgrade Using a TDM400, an ATA or ?? The phone works best with a 48V 20mA or better loop, so the FXS source voltage may have an effect, and this cheap phone has no previsions for external power. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SP3000 question
RumaTech wrote: Hi, all I finally got myself one of those SIPURA boxes. It is labeled as Linksys, but this is actually a SP3000 box. Anyway, unit has lots of configuration parameters. Not all are obvious. At the moment it registers against my *, but all the calls I do from analog phone connected to it, go to VoIP channel. As this part is still in testing, I want all the outgoing calls got to PSTN by default and dial, say 0, to get an outside VoIP line. I would like to do it as part of SP3000 configuration, not as part of * dialplan. Can someone help me? Use the gw0 directive in your SPA Dialplan. For example to get all 7 digit dialed calls to go out the FXO port, you can put in the following code: xxx:@gw0 Thanks, Rudolf P.S. I am reading the manual now, but it is rather large. I want to have SP3000 installed now, but to avoid screams from my wife, I want analog phone to work as an analog phone for now while I am playing with configuration and * dialplans. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SP3000 question
On Sat, Apr 22, 2006 at 11:19:35PM +1000, RumaTech scribbled: As this part is still in testing, I want all the outgoing calls got to PSTN by default and dial, say 0, to get an outside VoIP line. I would like to do it as part of SP3000 configuration, not as part of * dialplan. Can someone help me? I use the following dialplan within the Sipura: ([2-79]11:@gw0|999:@gw0|112:@gw0|0[12]x.|[*x]xx.:@gw0|#9,:[*x]x.|**) Using this, all emergency numbers go directly to PSTN, all numbers starting with 01 and 02 go via VoIP, and all other numbers go through PSTN. Any number prefixed with #9 is then forced to go through VoIP, with the initial #9 not being passed to Asterisk. Adapt and use. :-) Hope this helps, Roshan -- http://roshan.info Be different, act normal. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SP3000 question
On 4/23/06, Roshan Sembacuttiaratchy [EMAIL PROTECTED] wrote: I use the following dialplan within the Sipura: ([2-79]11:@gw0|999:@gw0|112:@gw0|0[12]x.|[*x]xx.:@gw0|#9,:[*x]x.|**) [..snip..] Is this @stuff something new in the SPA3000 dialplan syntax? I have SPA-200x ATAs and I never saw any mention of this in the manual, which makes sense if it's a SPA3k new dialplan feature. Cheers, Gonzalo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3
Thanks for the response, I'll ask the client to change batteries, though it is a new phone less than two weeks. is there any reason why the Lanline(Verizon) work and not the Asterisk? The only differences is the Asterisk, Linksys router and the DSL modem. One of these 3 should be interfering. -- Original message -- From: John Novack [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2. Replace the batteries! Alkaline only, replace every 6 months 1.2.3 is also defective for other reasons. Upgrade Using a TDM400, an ATA or ?? The phone works best with a 48V 20mA or better loop, so the FXS source voltage may have an effect, and this cheap phone has no previsions for external power. John Novack___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SP3000 question
Gonzalo Servat wrote: On 4/23/06, Roshan Sembacuttiaratchy [EMAIL PROTECTED] wrote: I use the following dialplan within the Sipura: ([2-79]11:@gw0|999:@gw0|112:@gw0|0[12]x.|[*x]xx.:@gw0|#9,:[*x]x.|**) [..snip..] Is this @stuff something new in the SPA3000 dialplan syntax? I have SPA-200x ATAs and I never saw any mention of this in the manual, which makes sense if it's a SPA3k new dialplan feature. That dialplan function has been around since v2 code for the spa3k, but using the gw0 and gw1 part of it only applies to the spa3k. The gw0 implies the physical fxo pstn port. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can I get a recording from a CD to my asterisk digital assistant
I got someone to record the messages we want for our auto-attendant menu on a CD. All I have to do not is to upload the files into the asterisk box, however the format is not recognized by the Asterisk box. Question 1) What formats should the sound file be, so I can upload it to my asterisk box? Thanks --Davi-Ann ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on FreeBSD + Passive ISDN BRI
Ok, from what I can see _NO_ passive ISDN cards will work with Asterisk on freebsd, is this correct is it likely to change soon? Secondly, if this is likely to be the way for a while, what is the lease expensive card that will work with FreeBSD? Also, can I use DID (Direct Inward Dialling) on FreeBSD? Thanks for all your help to date. Regards, Cian Hughes ___ [EMAIL PROTECTED] mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-isdn To unsubscribe, send any mail to [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I get a recording from a CD to my asterisk digital assistant
You will need them in one of asterisk supported formats. wav, slin,gsm, g729, g723... Davi-Ann escribió: I got someone to record the messages we want for our auto-attendant menu on a CD. All I have to do not is to upload the files into the asterisk box, however the format is not recognized by the Asterisk box. Question 1) What formats should the sound file be, so I can upload it to my asterisk box? Thanks --Davi-Ann ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3
The book states "batteries not supplied" so perhaps they were never installed? And what FXS circuit are you using to interface to Asterisk? The difference in loop current between VeriZon and the local interface could be an answer John Novack [EMAIL PROTECTED] wrote: Thanks for the response, I'll ask the client to change batteries, though it is a new phone less than two weeks. is there any reason why the Lanline(Verizon) work and not the Asterisk? The only differences is the Asterisk, Linksys router and the DSL modem. One of these 3 should be interfering. -- Original message -- From: John Novack [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2. Replace the batteries! Alkaline only, replace every 6 months 1.2.3 is also defective for other reasons. Upgrade Using a TDM400, an ATA or ?? The phone works best with a 48V 20mA or better loop, so the FXS source voltage may have an effect, and this cheap phone has no previsions for external power. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I get a recording from a CD to myasterisk digital assistant
Is there any special encoding that I have to use? - Original Message - From: Alberto Sagredo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 22, 2006 4:15 PM Subject: Re: [Asterisk-Users] How can I get a recording from a CD to myasterisk digital assistant You will need them in one of asterisk supported formats. wav, slin,gsm, g729, g723... Davi-Ann escribió: I got someone to record the messages we want for our auto-attendant menu on a CD. All I have to do not is to upload the files into the asterisk box, however the format is not recognized by the Asterisk box. Question 1) What formats should the sound file be, so I can upload it to my asterisk box? Thanks --Davi-Ann ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: SPA 3000 - UK Replacement
tom wrote: You think that's bad, I ordered one on the 10th of march from redstore, that was showing a 3-5 day. They still haven't despatched the unit and I have been trying to call them now (on their 0870 number) for about a week, during the past 3 weeks I have been sending them email after email that hasn't been responded to. Hiya! I had that too with RedStore. The order tracking was saying for absolutely AGES that it was waiting to come into stock. I did manage to get to speak to someone and was assured that they were awaiting delivery. Eventually (took about a month (or two??)) it turned up! - Had it now up and running since March and works fine (after figuring out that I needed Mod Taps to hook a phone into it to make it work!) admittedly RedStore did give me the option to cancel the order - but I stuck with it as it was nearly half the cost from anywhere else (about £50). Like I say though - this was about 6-8 weeks or so ago since I took delivery - I haven't checked to see if they are still selling them. Wayne. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SP3000 question
Hiyall, I don't suppose anyone has the elusive 'administrators' manual for these things - I've got the users manual but would still like the full suit so to speak. Cheers Wayne. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ?
The telco guys probably did something non-industry standard and reversed send and receive in the jack that they plugged the CAT5 into. Sure it works, sure it is easier, sure it is not the correct way of doing things. Thanks, Steve From: [EMAIL PROTECTED] on behalf of Lacy Moore - Aspendora Sent: Sat 4/22/2006 2:55 PM To: Paul Mahler; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ? att (formerly SBC, formerly Southwestern Bell, formerly ATT) just came out and installed my PRI. FYI, they used Cat 5e cable. No special T1 cabling that costs a fortune to buy somewhere, just plain old Cat 5e cable. Guess what guys? If they are using this as customers' sites, they are probably using it elsewhere. It's only as good as the weakest link, so you can go out and spend lots of money on T1 cable, or just use Cat 5e like the telco guys do. -- Lacy Moore Aspendora, Inc. winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?
I have used cross-connect wire from the spool to make T1 crossover cables with RJ45 ends. All that matters is that pin one goes to four and two goes to five on both ends. From: [EMAIL PROTECTED] on behalf of Andrew Sent: Sat 4/22/2006 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ? Alexander Lopez wrote: I have not in my experience seen any problems with using a Good Quality Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you should be fine. As far as the shielding goes, I use UTP cables and Connectors all the time and some of my X-connects run over 100 feet I have used cat-5 for everything communications. serial printers, dumb terminals, DS1 and even 10/100 ethernet. :-) It's easier to have it installed as a network jack and then use for whatever you need. ... Andrew McRory LinuxSystems Tallahasse, FL ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Don't see my post
Gafachi can, I've been using them with you problems. -- Original message -- From: [EMAIL PROTECTED] First of all, try sending it to the asterisk-biz list. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John RichSent: Monday, April 17, 2006 10:53 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Don't see my post Hi Folks,I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes.ThanksJohn. Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. ---BeginMessage--- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to restrict simultaneous phone registrations
disable three-way calling, restric channels to one per call. -- Original message -- From: "Bill Gibbs" [EMAIL PROTECTED] I say just bill the user at extension 333 it's his responsibility to keep the login info private. If he disputes it, refund the first time then change the password to something really complicated then start billing him if it keeps happening after that! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bryan Mahin Sent: Wednesday, April 05, 2006 10:50 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] How to restrict simultaneous phone registrations :) I should rephrase my question. And included a bit more information on wha t I am trying to accomplish. Solution 1 (preferred) I am working on an asterisk installation where most end users will use softphones. If I am not able to lock down calling to one phone at a time, the end users will share their login information with friends, family, neighbors, and the some girl they meet on myspace. Currently, I am able to register two phones at separate locations with the same account on each phone and make concurrent calls. For example, If I login extension 333 at location A, and 333 at location B, simultaneous calls can be placed from both phones at the exact same time. I only want calls placed from extension 333 to work from either A or B not A and B concurrently. Here is my ideal solution. Location A wants to make a call, but location B has a call in progress. Location B has to either close their pho ne, or hang up before Location A can make the call.OR.. Solution 2. :) A way I can distinguish in my CDR the IP address or some other recognizable difference between the two locations when they make concurrent calls using the same extension. The complication here is; I can currently the log IP addresses, but as the end phones are on the internet, Nat'd, and I am using a siparator for traversal. As a result, my logs show the IP address of the siparator and I don't have any other data to distinguish the end phones. OR.. Solution 2.5 One thought I've had is to send logs from the siparator to a syslog server, parse them, hunt for simultaneous calls placed by the same accounts from different locations, and bill the end users accordingly. But I really dislike this idea as no one likes to be hit with surcharges. ; Any help or insight is greatly appreciated. Thanks again, Bryan Mahin-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric "ManxPower" Wieling Sent: Wednesday, April 05, 2006 7:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to restrict simultaneous phone registrations Bryan Mahin wrote: Hello all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solut ion fo r this type scenario? This is a non-issue, because a second registration to the same account will override and previous registrations for that account. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please visit us @ www.uneta.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or up date options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No DTMF
I had the same problem, I reloaded Asterisk 1.2.3 and set the dtmf 2833 that fixed it. -- Original message -- From: "Mark Edwards" [EMAIL PROTECTED] Try dtmfmode=info and see if that works. Mark -Original Message-From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Thursday, 9 March 2006 6:08 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] No DTMF Some one was on my server making changes to my sip.conf files. I am now having trouble with DTMF. No matter what I use (inband,auto,rfc2833) the dtmf tones seem to not come thru. I compared it to the wiki and all the configs seem to be in order. Here is my sip.conf [general]disallow=all;allow=g729 ; requires license for g729allow=ulawport = 5060nat=yescontext=from-sipbindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)maxexpirey=4800 ; Maximum expiration for registrationsdefaultexpirey=1800 ; Default expiration for registrationscanreinvite=no ; Allow clients to directly connect if set to yes. Set to no if behind NAT.tos=reliabilitysrvlookup=yes ; Enable DNS SRV lookups on outbound callsvideosupport=no ; Turn on support for SIP videodtmfmode=rfc2833 ;rfc2833 ;inband ;rfc2833 ; DTMF inband need to be set here.pedantic=noexternip=..XXX ;Sip Mediaregister = XX:[EMAIL PROTECTED]/7322761368 [sipmedia6]type=frienduser=XX ;(Phone Number)username=XX ;(Phone Number)fromuser=XX ;(Phone Number)authname=XX ;(Phone Number)secret= ;(SIP Password)host=sip.sipmedia.com disallow=allallow=ulawcontext=ServerHighwayrealm=sip1.xchangetele.comfromdomain=sip.sipmedia.comdtmfmode=rfc2833canreinvite=no insecure=very Here is my extensions.conf [general]static=yeswriteprotect=yes [ServerHighway];Play Server Highway IVR Exten = s,1,Background(server-highway-ivr)Exten = s,2,Background(blank-file-10) Exten = 1,1,Ringing()Exten = 1,2,Wait(15)Exten = 1,3,Macro(stdexten,9511,9511)Exten = 2,1,Ringing()Exten = 2,2,Wait(15)Exten = 2,3,Macro(stdexten,9512,9512)Exten = 3,1,Ringing()Exten = 3,2,Wait(15)Exten = 3,3,Macro(stdexten,9513,9513)Exten = 4,1,Ringing()Exten = 4,2,Wait(15)Exten = 4,3,Macro(stdexten,9514,9514)Exten = i,1,Background(invalid)Exten = i,2,Goto(s,1) Exten = t,1,Goto(s,1) exten = 9,1,Goto(s,1);Extension To Record Main IVR Messageexten = 500,1,Authenticate(XXX)exten = 500,2,Record(ServerHighwayIvr:gsm) Yahoo! MailBring photos to life! New PhotoMail makes sharing a breeze. ---BeginMessage--- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Don't see my post
Also, this is really a biz list question. From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED] Sent: Sat 4/22/2006 6:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Don't see my post Gafachi can, I've been using them with you problems. -- Original message -- From: [EMAIL PROTECTED] First of all, try sending it to the asterisk-biz list. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Rich Sent: Monday, April 17, 2006 10:53 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Don't see my post Hi Folks, I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes. Thanks John. Yahoo! Messenger with Voice. Make PC-to-Phone Calls http://us.rd.yahoo.com/mail_us/taglines/postman1/*http:/us.rd.yahoo.com/evt=39663/*http:/voice.yahoo.com to the US (and 30+ countries) for 2¢/min or less. winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need help with getting EXTEN from pstn hunt group
Hi I have a TDM card with 4 lines on a hunt group coming in. I can answer the phones with exten = s,1,Answer() exten = s,n,Dial(ZapZap) ... The problem is I don't know how to find out what extension was originally dialed. And, trying to match on the extension always fails. E.g. exten = 1234567,1,Answer() # never gets here I thought I could get the extension on the 's' extensions above, but, no, the extension is 's'. Is there something special that needs to be done with pstn hunt groups to get the extension? Thanks -- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help with getting EXTEN from pstn hunt group
Jim, You might want to be a little more specific: a. You want to find out which line the call came in on, OR b. The actual PSTN number that was dialed An example: - assuming a hunting pstn number 2341000 - 4 lines in a the group: 2341001, 2341002, 2341003, 2341004 - The 4 lines, are connected to your TDM card as Zap/1, Zap/2, Zap/3, Zap/4 If you want to find out which line was actually called, put each line into a different context in your zapata.conf, e.g. context=pstn_2341001 channel=1 context=pstn_2341002 channel=2 ... In your extension.conf, you'll need something like [psnt_2341001] exten = s,1,Answer() exten =s,2,Set(DNID,2341001) exten =s,3,Goto(defuault,s,1) ; Jump to normal processing [psnt_2341002] exten = s,1,Answer() exten =s,2,Set(DNID,2341002) exten =s,3,Goto(defuault,s,1); Jump to normal processing But, if you want to get the original hunting number 2341000, you'll need to use an ISDN line. Hope this helps. Leo Jim Freeze wrote: Hi I have a TDM card with 4 lines on a hunt group coming in. I can answer the phones with exten = s,1,Answer() exten = s,n,Dial(ZapZap) ... The problem is I don't know how to find out what extension was originally dialed. And, trying to match on the extension always fails. E.g. exten = 1234567,1,Answer() # never gets here I thought I could get the extension on the 's' extensions above, but, no, the extension is 's'. Is there something special that needs to be done with pstn hunt groups to get the extension? Thanks -- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unicall MFRC2 Problems with BrT.
Are you sure its from today? The file has dates libmfcr2-0.0.3.tar.gz 30-Mar-2006 09:06 346K Also inside th tar the changelog has nothing inside and the news file has nothing too. How did you see it was from today? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Moises Silva |Sent: Saturday, April 22, 2006 9:21 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT. | |hum, the last time i downloaded something every file has |different dates. However, im looking at a new version that i |have downloaded |today: | |http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre9/ libmfcr2-0.0.3.tar.gz | |And checking the source it seems that tar is the most recent version. |I check the version looking in the C code for a fix i know |must be there, in mfcr2.c line 2780, after the generation tone |it must OR the signal with 0x80. | |Let me tell you that I have not tested that version. I have a |custom version that i fixed (because it gave me the same error |you have) and I sent the fix to Steve Underwood, but he told |me that my fix was not error proof, and that may fail (I have |1 month now in a production server with no problems tough), so |he made a similar fix, and told me that was more reliable. The |link I just gave you is for the TAR with Steve Underwood fix. | |I guess you already contacted me off-list to quote you for my |consultory. If you still have problems let me know and i may |be able to help you through SSH. | |Best Regards | |On 4/21/06, Anton Krall [EMAIL PROTECTED] wrote: | Moises, how can I find out which version Im running, on |Steves ftp all | say | 0.0.3 or the date also says the same date. | | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf |Of Moises | |Silva | |Sent: Friday, April 21, 2006 9:43 AM | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT. | | | |A couple of weeks ago, libmfcr2 has a small error in the tone | |signaling for the call setup, that was fixed 2 weeks ago or so, | |please, wich version of libmfcr2 are you using? if you dont |know try | |upgrading to the latest version. Im pretty much sure that you have | |the very same problem we had. | | | |Regards | | | |On 4/21/06, Jefferson Carvalho [EMAIL PROTECTED] wrote: | | Hello All, | | | | I'm facing problems with Unicall on this scenario : | | | | CentOS 4.3 - Running on x86_64 | | Asterisk 1.2.7.1 | | Zaptel 1.2.5 | | | | When running zttool , shows all Spans OK. | | | | But I can't receive and make calls. | | | | I tried to change many parameters and still doesn't work. | | | | Any clues ? | | | | * unicall.conf | | | | [channels] | | | | language=br | | | | context=incoming-pstn | | usecallerid=yes | | hidecallerid=no | | immediate=no | | callwaitingcallerid=yes | | threewaycalling=yes | | transfer=yes | | cancellforward=yes | | callreturn=yes | | echocancel=yes | | echocancelwhenbridged=yes | | | | rxgain=0.0 | | txgain=0.0 | | faxdetect=both | | loglevel=255 | | protocolclass=mfcr2 | | protocolvariant=br,20,4 | | protocolend=cpe | | group=1 | | callgroup=1 | | | | channel = 1-15 | | channel = 17-31 | | channel = 32-46 | | channel = 48-62 | | channel = 63-77 | | channel = 94-108 | | channel = 110-124 | | | | * zaptel.conf * | | | | loadzone=br | | defaultzone=br | | | | | | span=1,1,0,cas,hdb3 | | cas=1-15:1101 | | cas=17-31:1101 | | | | span=2,0,0,cas,hdb3 | | cas=32-46:1101 | | cas=48-62:1101 | | | | | | span=3,0,0,cas,hdb3 | | cas=63-77:1101 | | cas=79-93:1101 | | | | span=4,0,0,cas,hdb3 | | cas=94-108:1101 | | cas=110-124:1101 | | | | | | | | * lor error * | | | | -- Executing Dial(SIP/1000-1de2, | | Unicall/g1/40020022|40|Ttr) in new stack Apr 20 19:13:57 | | WARNING[30676]: chan_unicall.c:627 | | unicall_report: MFC/R2 | | UniCall/1 Call control(1) | | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: | | MFC/R2 | | UniCall/1 Make call | | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: | | MFC/R2 | | UniCall/1 Making a new call with CRN 32769 Apr 20 19:13:57 | | WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 | | UniCall/1 0001 - [1/ 1/Idle /Idle ] | | -- Called g1/40020022 | | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:2644 |handle_uc_event: | | Unicall/1 event Dialing | | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 | |unicall_report: MFC/R2 | | UniCall/1 - [1/ 40/Seize /Idle ] | | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 | |unicall_report: MFC/R2 | | UniCall/1 4 on - [2/ 40/Group I /Idle ] | | Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 | |unicall_report: MFC/R2 | | UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS | | ] cause | | 32769 - T1 timed out | |
