[Asterisk-Users] Re: Odd Zap dialing problem
I may have found a solution to my zap dialing problem: disabling the MMX option in the zaptel drivers. After disabling MMX and recompiling (I had dome some recompiles earlier to adjust tone length and such, with the problem still appearing), it's been over 36 hours since the last no-dial symptoms were seen. -- Seth et lux in tenebris lucet Mattinen [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 alist of codec by asterisk version?
Hi Il sab, 2004-07-10 alle 05:43, Kevin Walsh ha scritto: Joe Baptista [EMAIL PROTECTED] wrote: Is there alist of codecs asterisk actually has per version number - i.e. 0.7, 0.9 etc? if you have it installed, do show translation on the cli and you'll see all codecs supported, along with translation tims I believe Asterisk has the same codec list in all of its versions. Well, at least for the versions I've seen. I'm sure someone will rush to correct me if I'm wrong. as far as I know, the stable branch doesn't have g726 as in HEAD branch. matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] bristuff - hfc card + x100p
I got the everyone's busy at this moment too , I have one hfc isdn card and one x100p , it was because of me using one channel of my isdn for internet , so i just switched to second channel and it dialed out fine , you need to edit this in extensions.conf for outgoing dial out. You can also solved it in this way , i.e. if first channel is busy , use second. regards ~uppal On Fri, 09 Jul 2004 13:22:55 +0200, Tomaz [EMAIL PROTECTED] wrote: just one thing I forgot: less /proc/zaptel/3 Span 3: WCFXO/0 Wildcard X101P Board 1 RED 7 WCFXO/0/0 /proc/zaptel/3 (END) if this may helps? thankyou, Tomaz Tomaz wrote: hi, i have two hfc isdn cards and one x100p modules are loaded in order: - insmod zaptel insmod zaphfc modes=0 insmod wcfxo ztcfg cat /etc/zaptel.conf loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 # loadzone=nl defaultzone=nl span=2,1,3,ccs,ami bchan=4-5 dchan=6 # loadzone=nl defaultzone=nl fxsks=7 --- and in zapata.conf i have: [channels] switchtype = euroisdn signalling = bri_cpe prilocaldialplan=national pridialplan = unknown echocancel=yes echocancelwhenbridged=yes ;echotrainig=yes overlapdial=no immediate=no group = 3 context=isdn channel = 1-2 ; switchtype = euroisdn signalling = bri_cpe prilocaldialplan=national pridialplan = unknown echocancel=yes echocancelwhenbridged=yes ;echotrainig=yes overlapdial=no immediate=no group = 3 context=isdn channel = 4-5 ; signalling=fxs_ks channel=7 so after that i load asterisk -vc and get no errors at all: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Registered channel 4, PRI Signalling signalling -- Registered channel 5, PRI Signalling signalling -- Registered channel 7, FXS Kewlstart signalling == Starting D-Channel on span 1 == Starting D-Channel on span 2 == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) isdn is working ok but with channel 7 (x100p) is something wrong .. but what? Executing Dial(SIP/1-85f5, Zap/7/BYEXTENSION|130|t|r|f) in new stack Jul 9 14:43:32 NOTICE[311316]: app_dial.c:559 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time thankyou! Tomaz Robinson Tim-W10277 wrote: Yes, me. What problems are you having? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomaz Sent: 09 July 2004 09:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] bristuff - hfc card + x100p hi! Anyone on list has working asterisk box with hfc based card (bristuff) and a x100p adapter? Becouse together in box I can't get it working in any way .. thank you, Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 Support for ISDN National-3?
Has anyone tried to run Asterisk connected to a Central Office using an ISDN National ISDN 3 PRI trunk? George Pajari netVOICE communications www.netvoice.ca www.ip-centrex.ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to differentiate a *busy* call from not available?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello On 10/07/2004, at 5:45 AM, Soren Rathje wrote: Based on extensions.conf.sample from CVS-HEAD... Thank you very much for this information ; for some reason I can't seem to get other version newer than June,29th As a side note, it's pretty amazing that people complain I could ask question on a new feature that was added only a few weeks ago... Anyway.. I got something working now thanks to your information, as well as with call forwarding. There's just what thing I can't figure out. What is the action for s-. Thank you Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA79HUXeDVKqIr3GURAgFSAKCBN2m5lwHOcVyvzFEZJyoz8GPNhACdES1X oYahc18y7RbK/wjeAHENYvc= =XxQB -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to differentiate a *busy* call from not available?
Jean-Yves Avenard wrote: There's just what thing I can't figure out. What is the action for s-. It's the better safe than sorry option... :-) Basically it's a wildcard option, anything beginning with s- will go there... -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] IVR Menu and VoiceMail quality
On 9 Jul 2004 at 14:08, Chris Shaw wrote: Thx Jay, I hope this is not a too FAQ... I really did try to look it up first but I saw s many conflicting things about timing... one person says no you absolutely do not need ztdummy or a digium card to make IVR/Voicemail work, others say you need it for everything... I tend to believe the latter since it seems to be more of a timing issue than a bandwidth issue... What I can't figure out though is if it's a timing thing, shouldn't calls on my local net be crappy too? When I log into voicemail from my ip phone it's perfect... when I call home from out of town it sounds like crap unless I play with the nice values or restart asterisk... Just a thought, when setting up your QOS, did you make sure that the maximum usage was slightly below your actual pipe size? Matt - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 09, 2004 1:48 PM Subject: RE: [Asterisk-Users] IVR Menu and VoiceMail quality AFAIK, it's needed anytime asterisk streams audio... Which is meetme, MOH and of course voicemail and IVR. My Asterisk system had lousy IVR quality until I plugged in the FXO card and loaded Zaptel. -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Friday, July 09, 2004 3:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IVR Menu and VoiceMail quality I thought it was only needed for MeetMe and MOH? - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 09, 2004 12:21 PM Subject: RE: [Asterisk-Users] IVR Menu and VoiceMail quality Do you have ztdummy loaded? -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Friday, July 09, 2004 1:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IVR Menu and VoiceMail quality I have really tried to do my best googling and wiki-reading before asking this question. I couldn't find the answers there so I throw myself at the mercy of the list... I get excellent quality for SIP - PSTN and PSTN - SIP calls, however when I or anyone else calls from PSTN - * the voice menus are oftentimes very choppy. Sometimes they are absolutely perfect and I cannot tell that it's actually VoIP. Sometimes it's so bad that I can't understand what Allison's saying at all... Calls on the same network sound just fine... I know what you're thinking, it's a congested link, and that may be but I've noticed that if I play with the nice value of asterisk, it seems to help. Setting nice to 0 seems to work the best, I tried -20 and it was the worst... I have implemented QoS on my network and have given any and all asterisk packets priority. As I said actual calls are crystal clear so I believe it to be a problem with Asterisk itself or the machine it's running on. Possibly some bottleneck somewhere. I realize that since it's going over the public internet, the occasional dropped packet is to be expected, but what's frusterating is that I can get crystal clear menus sometimes even when my network is fully loaded and other times when it's perfectly quiet it sounds absolutely horrible... Here are the machine's specs if that helps: AMD Athlon 1Ghz (Old Thunderbird core) Asus A7V600 128MB DDR-266 RAM 450GB storage (4 IDE drives in an LVM array) *grin* Pure VoIP, no digium hardware Internet connection is cable with 3Mbit downlink and 256Kbit uplink... As I said earlier I wouldn't have even asked, but it dosen't seem to be totally bandwidth related so I'm wondering if anyone has any ideas... Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bad clicking sounds with Diva+capi+asterisk
Hello, We have been using a Diva 4BRI with our Asterisk PBX through the capi interface for almost a year now with good results. However, recently we started to hear heavy clicking sounds in our phones when two simultaneous incoming calls are processed by the card. The clicking does not originate with the phones as it happens also in voicemail left directly on the server and happens with different phone models. Theses interferences almost prevent us from hearing the remote party and they don't always happen on all simultaneous calls. For instance if we hang up a corrupted call and call back then the line is clean. The clicking sometimes sounds like crosstalk between both lines. After talking on irc with chan_capi's developper (kapejod) it seems the problem might originate from the Diva card. We are using a vanilla 2.6.7 kernel on debian unstable. We recently updated to use the very latest Diva firwmare and divactrl-2.1 without any effect on the problem. I suspect the problem might be related to the Diva 4BRI drivers in kernel 2.6.x as the problem seems to have started when switching from 2.4.x. Has someone had a similar experience? Is there any solution? Thanks in advance for your insight, -- Jesus is coming! Everyone look busy! ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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)
Hi, my setup: Client: Win/linux client running x-lite or linphone Server: debian running asterisk on connect, incomming works well but outgoing to POTS has a lot of bad sound (no, the mic is ok, using logitec usb headset). to ensure proper work, tried normal p2p, worx well the sound is nearly unandertandable, can anyone help? ICQ:321628 thnx Stefan ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] displaying call progress with SendText on a Snom
I've had the same idea about using SendeText for call call progress information. Unfortunately, snoms currently do not support SIP MESSAGEs. I have recently contacted their excellent support about this. They said that they would consider it. regards martin Brady Alleman wrote: Is there a list of phones (hard or soft) that support the Asterisk SendText or SendURL applications? I have been trying to make this work with a Snom 200 and a Cisco 7960, to display call progress information, such as which trunk a outbound call is routed to, but my attempts have been unsuccessful. The Snom claims to support SMS somehow, but searches reveal no useful information on how it is to be done. Any tips? Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] German Asterisk Site
Hello Asterisk Users, is there a good german site for asterisk? Moritz
Re: [Asterisk-Users] T1 Hardware Echo Can
Brent, Best think through what you're trying to do here. You have multiple choices on how to interface * to the traditional pstn world, including the x100p, tdm cards, multiple T1 interface types, external gateways, etc. The x100p and tdm cards have an internal 2-wire to 4-wire hybrid that is built into the chipsets on those cards. In order to use an external echo can with them, you'd have to change that to a 4-wire to 4-wire arrangement, which cannot be done. The chipsets weren't engineered for that. Since the majority of echo issues with these two cards are the result of the card's hybrid (and internal * echo can functions), simply disabling the echo can function and attempting to replace that with an external echo can box won't accomplish anything. (The problem is already existing in front of that external box.) If you install a T1 card and an external T1 mux (with fxo cards), the echo can function already exists within the mux and/or cards. Don't really need 'another' external echo can box unless you actually purchased a T1 mux that didn't have echo can in the first place (and they do exist). If you install a PRI (or related types of channelized T1 arrangements), you don't need an external echo can function as those interfaces are generally 4-wire to 4-wire interfaces already. If echo exists, its generally the result of other interfaces (located somewhere else) and those locations should be addressing the corrective actions needed to resolve the issue. If you only need a small number of fxo ports, pstn gateways in the form of Cisco boxes, Mediatrix 1204, as well as many other products can do that at a relatively inexpensive cost. The echo can function is built into those boxes, which do work. Each comes with additional considerations or issues however. Lots of choices - Original Message - After reading the lists and taking reccomendations from TC, I have finally given up on the echo can built into asterisk. I am sick of hearing complaints from users, so the money spent on a hardware echo can will be worth its weight in gold. I am curious however, about some setup and component requirements. It seems as if every telecom place I call, either never calls back or doesn't have a clue what I am talking about. Does anyone have any good companies with competent sales/engineer people who would help put together a solution. Also, for anyone that has hooked up a echo can before. Do you have to buy such a large shelf? Obviously things things are intended for ILEC installs, however, I can't find anything geared towards the PBX realm. It seems everything on Ebay is a 32 module shelf rack. Thats a bit over kill for us. Further, I would imagine this setup. Please correct me if I am off base. I would have a straight T1 cable from the Channel Bank to the Echo Can, and then a X-over from * to the Echo Can. How are these solutions (e.g. Tellabs) wired up? Are there RJ45 connectors on the back of the shelf, or is it a strip the wire and twist method? Any assistance that anyone can provide to myself (and the list) would be greatly appreciated, as there are many people who would benefit from this... Thanks! - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X101P FXO with RED alarm
Hi, I've just added an X101P FXO card to my asterisk machine. The card is detected ok but zttool always shows it in a RED alarm state, which I understand means that it doesn't detect the line. I've connected the card via the line socket to my analogue line using the cable that came with the card (and have also tried other cables too) but I always get a RED alarm. I'm located in the UK, running FreeBSD-5.2, with zaptel-0.5 drivers and asterisk-0.9.0_1 (both built from ports). My zaptel.conf: fxsks=1 loadzone=uk defaultzone=uk My zapata.conf [channels] signalling=fxs_ks context=default channel = 1 Output of ztcfg -vv: Keyword: [fxsks], Value: [1] Keyword: [loadzone], Value: [uk] Keyword: [defaultzone], Value: [uk] Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Asterisk sees the zap channel (or at least it shows up with 'zap show channels') but always sees it as busy, presumably because of the red alarm. The other thing of note is that when the zaptel driver is loaded, attempting to play MOH causes the entire system to lock up and require a reboot. (Without the driver loaded it works, but is of course choppy because of the lack of a timing source). Anyone have any ideas about what's going on? Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] T1 Hardware Echo Can
Rich Adamson wrote: [...] If you install a T1 card and an external T1 mux (with fxo cards), the echo can function already exists within the mux and/or cards. Don't really need 'another' external echo can box unless you actually purchased a T1 mux that didn't have echo can in the first place (and they do exist). [...] T1 muxes do not normally contain an echo canceller. It is a special item if it actually *does* contain an echo canceller. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] T1 Hardware Echo Can
Hi Rich, Thanks for your heads up. See comments below. From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rich Adamson Best think through what you're trying to do here. You have multiple choices on how to interface * to the traditional pstn world, including the x100p, tdm cards, multiple T1 interface types, external gateways, etc. Currently, I have a T100P card with a T1 crossover cable connected to an Adtran Total Access 750. We did have an Adit 600, but had a lot of problems with it (static, etc) and thought it was the channel bank. Long story short, ended up not getting along with Carrier Access, returned it and bought Adtran. Adtran CS is much friendlier from my expierences. So therefore, we are stuck w/ a CB without impedance matching :( The x100p and tdm cards have an internal 2-wire to 4-wire hybrid that is built into the chipsets on those cards. In order to use an external echo can with them, you'd have to change that to a 4-wire to 4-wire arrangement, which cannot be done. The chipsets weren't engineered for that. This shouldn't be an issue here. Our X100P's on a smaller branch office do not have that much echo, and we don't hear complaints from these users. If you install a T1 card and an external T1 mux (with fxo cards), the echo can function already exists within the mux and/or cards. Don't really need 'another' external echo can box unless you actually purchased a T1 mux that didn't have echo can in the first place (and they do exist). Our Adtran does not have an echo can built in. I couldn't find any Channel Banks that have this feature. Do you know of any? If you install a PRI (or related types of channelized T1 arrangements), you don't need an external echo can function as those interfaces are generally 4-wire to 4-wire interfaces already. If echo exists, its generally the result of other interfaces (located somewhere else) and those locations should be addressing the corrective actions needed to resolve the issue. I wish we could afford a PRI, however it is really cost prohibitive for us. If you only need a small number of fxo ports, pstn gateways in the form of Cisco boxes, Mediatrix 1204, as well as many other products can do that at a relatively inexpensive cost. The echo can function is built into those boxes, which do work. Each comes with additional considerations or issues however. The mediatrix boxes do look good. I'll have to consider for next project. Thanks for info! - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 + g.726
How I can do to use the g.726 on asterisk ? Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] T1 Hardware Echo Can
On Saturday 10 July 2004 11:21, Rich Adamson wrote: If you install a T1 card and an external T1 mux (with fxo cards), the echo can function already exists within the mux and/or cards. Don't really need 'another' external echo can box unless you actually purchased a T1 mux that didn't have echo can in the first place (and they do exist). As Steve already said, generally speaking echo cancellation hardware on T1/E1 interfaces is an option adder. If it doesn't mention it, it doesn't have it. If you install a PRI (or related types of channelized T1 arrangements), you don't need an external echo can function as those interfaces are generally 4-wire to 4-wire interfaces already. If echo exists, its generally the result of other interfaces (located somewhere else) and those locations should be addressing the corrective actions needed to resolve the issue. You will often hear echo from the far-end hybrid, even on PRI, as I have found out. Normal KSU/PBX systems with T1/PRI interfaces have echo cancellation hardware within the KSU itself. I am purchasing a T1 echo canceller in order to try and eliminate the echo we hear (i.e. far-end echo) -- something I didn't think I'd need to do. Our telco (Bell Canada) seems oblivious to any knowlege about echo cancellation for T1 within the CO, but I continue to press, because every now and again you hear a glimpse of oh yeah we can do that, your line just wasn't engineered with that equipment kind of blurb. :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: ATA 186, firmware SIP 3.1 and codec g.726 + now SIPURA SPA-2000
On Fri, 2004-07-09 at 13:55, [EMAIL PROTECTED] wrote: To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. It looks to me that you are using CVS -stable (which seems to support G726 PASSTHRU) and not CVS -head (which supports G726 TRANSCODING, which is what you need). What does show version at the CLI show. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] X101P FXO with RED alarm
You may need the 3-wire (UK) to 2-wire (US) adapters for the jack the X100P is plugged into. I've heard you can get them most anywhere. On Sat, 2004-07-10 at 09:40, Richard Airlie wrote: Hi, I've just added an X101P FXO card to my asterisk machine. The card is detected ok but zttool always shows it in a RED alarm state, which I understand means that it doesn't detect the line. I've connected the card via the line socket to my analogue line using the cable that came with the card (and have also tried other cables too) but I always get a RED alarm. I'm located in the UK, running FreeBSD-5.2, with zaptel-0.5 drivers and asterisk-0.9.0_1 (both built from ports). My zaptel.conf: fxsks=1 loadzone=uk defaultzone=uk My zapata.conf [channels] signalling=fxs_ks context=default channel = 1 Output of ztcfg -vv: Keyword: [fxsks], Value: [1] Keyword: [loadzone], Value: [uk] Keyword: [defaultzone], Value: [uk] Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Asterisk sees the zap channel (or at least it shows up with 'zap show channels') but always sees it as busy, presumably because of the red alarm. The other thing of note is that when the zaptel driver is loaded, attempting to play MOH causes the entire system to lock up and require a reboot. (Without the driver loaded it works, but is of course choppy because of the lack of a timing source). Anyone have any ideas about what's going on? Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] T1 Hardware Echo Can
Andrew Kohlsmith wrote: On Saturday 10 July 2004 11:21, Rich Adamson wrote: If you install a T1 card and an external T1 mux (with fxo cards), the echo can function already exists within the mux and/or cards. Don't really need 'another' external echo can box unless you actually purchased a T1 mux that didn't have echo can in the first place (and they do exist). As Steve already said, generally speaking echo cancellation hardware on T1/E1 interfaces is an option adder. If it doesn't mention it, it doesn't have it. If you install a PRI (or related types of channelized T1 arrangements), you don't need an external echo can function as those interfaces are generally 4-wire to 4-wire interfaces already. If echo exists, its generally the result of other interfaces (located somewhere else) and those locations should be addressing the corrective actions needed to resolve the issue. You will often hear echo from the far-end hybrid, even on PRI, as I have found out. Normal KSU/PBX systems with T1/PRI interfaces have echo cancellation hardware within the KSU itself. I am purchasing a T1 echo canceller in order to try and eliminate the echo we hear (i.e. far-end echo) -- something I didn't think I'd need to do. Our telco (Bell Canada) seems oblivious to any knowlege about echo cancellation for T1 within the CO, but I continue to press, because every now and again you hear a glimpse of oh yeah we can do that, your line just wasn't engineered with that equipment kind of blurb. :-) I think you are missing something important about how traditional telephone networks function. In the days before echo cancellation was practical, it was vital to avoid the need for them. They couldn't avoid the echo, so they avoided significant delays. Within almost any country, the physical delay is so short the echo from the far end appears as pleasant reverberation, and not nasty echo. International circuits have always been a pain, as significant delay is unavoidable there. It is packetising voice that really introduced delay as a broad issue. First in digital cellular networks, where codecs process voice in blocks, and inherently introduce at least a one block (say 20ms) delay. Now VoIP broadens the issue further. Equipment makers specifically design traditional network equipment to minimise delay. When I was developing DSP processing within the PCM network I was only allowed 375us (3 samples) delay - one sample to de-serialise the PCM stream, one to process it, and one to re-serialise the result. Delay budgets are always set as tight as possible. Bottom line: the traditional PSTN has always had echo, and it is normally irrelevant. Telcos, have no need and no interest in removing it. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 + g.726
On Sat, 2004-07-10 at 10:11, [EMAIL PROTECTED] wrote: How I can do to use the g.726 on asterisk ? Use Asterisk CVS -head. http://www.asterisk.org/index.php?menu=download --Eric -- Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the Unofficial Links) and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] strange echo problem
We have a strange echo problem. Maybe echo isn't the correct term. When we make a call f/ a SIP phone (we have several 7960's, some 3coms, and I've even tried a softphone, all on the same 100BaseTX network) to the pstn, if the person I'm calling has a PRI or channelized T1 f/ Bell, then the sound is perfect, couldn't be better. If I make a call to a person with a plain POTS line, I hear everything I say in my earpiece about 1/4 second after I say it. It's very irritating.We have tried 2 different * boxes, using 2 different T1/PRI cards f/ digium. After calling digium about it, we set echotraining to 800 in zapata.conf. It got better but was still there, if I turn the volume down on the phone, it does almost go away, but it's still detectable. No where near as clear as calling a person that has a PRI or channelized T1 for phone service. The POTS persons we call that we do have the echo issue with all say the call sounds perfecto to them. Am I missing something obvious? We experience the exact same issue, and like Rich said in a subsequent post, I'm thinking there's a gremlin hiding somewhere in the * code. Everyone said you shouldn't have to even use echo canceling on a T1 PRI, but we do or we get serious complaints, instead of consistent minor complaints. FWIW, it still was around in the 6/29 CVS and we just updated again last night. For us the echo is a slight faint echo now that we implemented the echotraining=800, but it's still there. We haven't touched TX/RX gain. We can also give anyone access and a SIP account if that would be helpful. Best regards, Ryan Thrash ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] T1 Hardware Echo Can
If you install a T1 card and an external T1 mux (with fxo cards), the echo can function already exists within the mux and/or cards. Don't really need 'another' external echo can box unless you actually purchased a T1 mux that didn't have echo can in the first place (and they do exist). Just to be clear here there are a lot of channel banks, fact most that are suggested here (ta 750, ab i/ii, adit 600 and zhone etc) that dont have have echo cans that do benefit from the hardware echo cans eg ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 Book
Not from me. I think the more books the better. I'm looking forward to getting my copy. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Bailey Sent: Friday, July 09, 2004 2:52 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Book Hello, If anyone is interested in getting a book on asterisk I would recommend checking out http://www.saww.net/asterisk/ I ordered a copy, but they said it's six weeks or so 'till delivery. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training Do I detect some friendly rivalry? ;-) | VoIP Telephony with Asterisk will be available July 22, directly from | Signate and through selected resellers for $49.95 plus shipping. Call | 415-442-4011 to order the book. Seriously, though, the more documentation the better. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] T1 Hardware Echo Can
Steve Underwood wrote: Andrew Kohlsmith wrote: On Saturday 10 July 2004 11:21, Rich Adamson wrote: If you install a T1 card and an external T1 mux (with fxo cards), the echo can function already exists within the mux and/or cards. Don't really need 'another' external echo can box unless you actually purchased a T1 mux that didn't have echo can in the first place (and they do exist). As Steve already said, generally speaking echo cancellation hardware on T1/E1 interfaces is an option adder. If it doesn't mention it, it doesn't have it. If you install a PRI (or related types of channelized T1 arrangements), you don't need an external echo can function as those interfaces are generally 4-wire to 4-wire interfaces already. If echo exists, its generally the result of other interfaces (located somewhere else) and those locations should be addressing the corrective actions needed to resolve the issue. You will often hear echo from the far-end hybrid, even on PRI, as I have found out. Normal KSU/PBX systems with T1/PRI interfaces have echo cancellation hardware within the KSU itself. I am purchasing a T1 echo canceller in order to try and eliminate the echo we hear (i.e. far-end echo) -- something I didn't think I'd need to do. Our telco (Bell Canada) seems oblivious to any knowlege about echo cancellation for T1 within the CO, but I continue to press, because every now and again you hear a glimpse of oh yeah we can do that, your line just wasn't engineered with that equipment kind of blurb. :-) I think you are missing something important about how traditional telephone networks function. In the days before echo cancellation was practical, it was vital to avoid the need for them. They couldn't avoid the echo, so they avoided significant delays. Within almost any country, the physical delay is so short the echo from the far end appears as pleasant reverberation, and not nasty echo. International circuits have always been a pain, as significant delay is unavoidable there. It is packetising voice that really introduced delay as a broad issue. First in digital cellular networks, where codecs process voice in blocks, and inherently introduce at least a one block (say 20ms) delay. Now VoIP broadens the issue further. Whoops. Codecs introduce an inherent 2 block delay - one to compress and one to decompress. With most codecs that could be fudged a little to only introduce about a 1.5 block delay, but never seems to be in practice. Equipment makers specifically design traditional network equipment to minimise delay. When I was developing DSP processing within the PCM network I was only allowed 375us (3 samples) delay - one sample to de-serialise the PCM stream, one to process it, and one to re-serialise the result. Delay budgets are always set as tight as possible. Bottom line: the traditional PSTN has always had echo, and it is normally irrelevant. Telcos, have no need and no interest in removing it. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] German Asterisk Site
Beierlein Moritz wrote: Hello Asterisk Users, is there a good german site for asterisk? Moritz Hi Moritz, there is * dicussion group at the German IP-Phone forum: http://www.ip-phone-forum.de/ http://www.ip-phone-forum.de/forum/viewforum.php?f=24 jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] X101P FXO with RED alarm
Richard Airlie [EMAIL PROTECTED] wrote: I've just added an X101P FXO card to my asterisk machine. The card is detected ok but zttool always shows it in a RED alarm state, which I understand means that it doesn't detect the line. I've connected the card via the line socket to my analogue line using the cable that came with the card (and have also tried other cables too) but I always get a RED alarm. I'm located in the UK, running FreeBSD-5.2, with zaptel-0.5 drivers and asterisk-0.9.0_1 (both built from ports). First things first. Scrap the ports and build from the latest CVS source. 0.9 is far to old and buggy, and suspect the same of the Zaptel driver you have, although I don't use *BSD myself. Secondly, the red alarm does tend to mean that the line is not connected, but I got what you're describing when I moved Asterisk to a new machine. Try the X100P card in a different PCI slot. That cleared it for me, for whatever reason. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] X101P FXO with RED alarm
Eric Wieling [EMAIL PROTECTED] wrote: You may need the 3-wire (UK) to 2-wire (US) adapters for the jack the X100P is plugged into. I've heard you can get them most anywhere. No - steal the RJ-BT cord off the back of a modem if the X100P didn't come with a BT plug. It'll work without any adapters. Most PC shops will sell you a RJ-BT cord - or try Maplin/Tandy etc. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] German Asterisk Site
Das is aber schöen! Paul von Wachter Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jo Sent: Saturday, July 10, 2004 8:38 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] German Asterisk Site Beierlein Moritz wrote: Hello Asterisk Users, is there a good german site for asterisk? Moritz Hi Moritz, there is * dicussion group at the German IP-Phone forum: http://www.ip-phone-forum.de/ http://www.ip-phone-forum.de/forum/viewforum.php?f=24 jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Three (quick?) questions...
[Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones inside? [In other words, I obviously need a NIC -- but what would be the bare-bones telco POTS interface?] 2) What phones would be recommended for inexpensive (doesn't even need LCD), and yet functional? 3) In order to share data and voice over a T1, does it have to be PRI? [I've got a T1 I could probably play with, but I'd like to be sure it'll... well, you know: work.] Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: Problem of loading the oh-323 module
Remarque : message transféré en pièce jointe. Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !Téléchargez Yahoo! Messenger sur http://fr.messenger.yahoo.com---BeginMessage--- Remarque : message transféré en pièce jointe. Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !Téléchargez Yahoo! Messenger sur http://fr.messenger.yahoo.com---BeginMessage--- Hello everybody, I am still working on Asterisk, everything worked fine untill now, but now my problem is in loading oh323 module by Asterisk: The error that I have is : [chan_oh323.so]Jul 9 14:11:11 WARNING[1076298368]: loader.c:242 ast_load_resource: /usr/local/lib/liboh323wrap.so: undefined symbol: _ZTI14PAbstractArray Jul 9 14:11:11 WARNING[1076298368]: loader.c:374 load_modules: Loading module chan_oh323.so failed! I am using the following versions: asterisk-oh323-0.6.3a openh323-v1_13_5-src pwlib-v1_6_6-src CVS HEAD sources of Asterisk (29/06/2004) Before, I had older versions of asterisk-oh323, openh323 and pwlib, so I had a lot of problems in compiling the different elements, and now that everything compile, I cannot load the oh323 module...maybe should I upgrade aserisk, but if it is the case should I recompile the others again? I am afraid of having bad surprises... Can someone help me urgently please?? Thank you very much Soumaya Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !Téléchargez Yahoo! Messenger sur http://fr.messenger.yahoo.com ---End Message--- ---End Message---
[Asterisk-Users] How to use freeradius for Asterisk billing
Hello everybody, Can someone help me to find the right elements and the right configuration to send the CDRs of Asterisk to a freeradius servers? I found some help from the page http://bugs.digium.com/bug_view_page.php?bug_id=0001193 but It didnt work for me... should I change something in res_radius.c? if yes what exactly? and what should I add in the dialplan to activate the radius accounting? I would be very grateful if somebody can help me... Best regards, Soumaya Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !Téléchargez Yahoo! Messenger sur http://fr.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Three (quick?) questions...
Ken D'Ambrosio wrote: [Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones inside? [In other words, I obviously need a NIC -- but what would be the bare-bones telco POTS interface?] X100P/X101P... but people don't like it, so take the little more expensive TDM400P with a single FXO interface. 2) What phones would be recommended for inexpensive (doesn't even need LCD), and yet functional? Some people like budgetone's, but the site www.voip-info.org should reveal more information... If you want something cheaper... you can always get a second FXS module for the TDM400P and plug a standard analogue phone in it, maybe use one that supports FSK signalling for CallerID number + name. Even if you won't use this in production, a TDM400P with both FXS and FXO interface is very nice for testing stuff in combination with Asterisk. 3) In order to share data and voice over a T1, does it have to be PRI? [I've got a T1 I could probably play with, but I'd like to be sure it'll... well, you know: work.] Yep, T100P should do in that case. Call Digium sales for details. Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Three (quick?) questions...
1) A digium FXO card ($100) will do. Works great for me, ymmv. ~$100. My system is running on a Celeron 2.7GGhz with 256 of RAM and the processor never really blips. I'd say a ~1 GHz would be plenty enough for one or two channels. 2) $10 Walmart Special connected to a $100 Sipura SPA-2000. The Sipura gives you 2 FXS ports (=two extensions) bringing the cost of an analog port to $50/each. Any analog phone will do... I have two cordless, two SWB Freedom Phones ($9.95, with caller-id) and a couple of Aastra 390s (refurbed from ebay, $40ish) because of their excellent speakerphone. 3) Dunno, I do POTS and VOIP only. -Original Message- From: Ken D'Ambrosio [mailto:[EMAIL PROTECTED] Sent: Saturday, July 10, 2004 11:33 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Three (quick?) questions... [Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones inside? [In other words, I obviously need a NIC -- but what would be the bare-bones telco POTS interface?] 2) What phones would be recommended for inexpensive (doesn't even need LCD), and yet functional? 3) In order to share data and voice over a T1, does it have to be PRI? [I've got a T1 I could probably play with, but I'd like to be sure it'll... well, you know: work.] Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] using asterisk voicemail with a class 5 softswitch
I am using asterisk as a voicemail server for our IP Centrex SoftPBX. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten Sent: 09 July 2004 22:46 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] using asterisk voicemail with a class 5 softswitch when you say you have integration what exactly do you mean? are you using asterisk as the voicemail system for a class 5 switch? On Friday 09 July 2004 15:45, usedcanon wrote: I have integration. Asterisk is upto the task however you may need to do some work arounds. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten Sent: 09 July 2004 20:51 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] using asterisk voicemail with a class 5 softswitch anyone have any idea on the compatibility of asterisk voicemail with a class 5 switch that can do SIP (in particular the MetaSwitch VP3500)? -- Chad Whitten Network/Systems Administrator [EMAIL PROTECTED] 601-944-4801 Phone ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Chad Whitten Network/Systems Administrator [EMAIL PROTECTED] 601-944-4801 Phone ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does the SPA-3000 get rid of echo that the X100P can't?
After trying everything under the sun to get rid of echo on my X100P, I'm curious if anyone managed to solve the echo issues by switching to a SPA-3000? As well, if you have multiple SPA-3000's, can you create dial-out groups similar to the Dial(ZAP/g1) functionality? Thanks. -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Avoiding transcoding
How can one specify that codec selection should avoid transcoding if possible? The reason I ask is that when enum lookup succeeds and the destination only accepts ULAW, the various transcodings seems to garble the audio if GSM, ILBC, etc. are allowed. Cheers, Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Avoiding transcoding
Dr. Murphey: How can one specify that codec selection should avoid transcoding if possible? Asterisk only transcodes if the original audio stream needs to be connected either to (a) another audio stream or (b) an internal Asterisk function -- and the original audio stream is encoded in a format that is not compatible with the other audio stream or Asterisk function. The reason I ask is that when enum lookup succeeds and the destination only accepts ULAW, the various transcodings seems to garble the audio if GSM, ILBC, etc. are allowed. To illustrate the point above further, if you have a destination that only accepts ULAW, and an audio stream in GSM or ILBC, you must transcode. It is simply not an option since ULAW is not GSM, Asterisk has to convert (transcode) the ULAW samples into GSM samples and vice versa. Either I've misunderstood your question, or you have a different understanding of transcoding than most. Hope the above helps and apologies if I've completely missed your point. George Pajari www.netvoice.ca www.ip-centrex.ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Three (quick?) questions...
1) X100P from Digium. We currently have four of them in one machine (started with one, then added a second, then went all the way up to four) and they're working great. Put the X100P into a cheap PC (we're using a 1.7 Ghz Celeron system, I believe), add Linux and Asterisk, and you have a full-featured PBX for cheap. 2) Either put a TDM400P in your box and use up to 4 standard analog phones, or get something like the IAXy from Digium or a Grandstream HandyTone or BudgeTone. The Grandstream phones aren't the best, but they're probably the cheapest for SIP connectivity. A last option would be a software solution using IAX or SIP, but that requires another computer for each extension. 3) No experience with T1 here, but we'll probably need to look at something else when we eventually need to add more lines. -Zak ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] T1 Hardware Echo Can
Rich Adamson wrote: [...] If you install a T1 card and an external T1 mux (with fxo cards), the echo can function already exists within the mux and/or cards. Don't really need 'another' external echo can box unless you actually purchased a T1 mux that didn't have echo can in the first place (and they do exist). [...] T1 muxes do not normally contain an echo canceller. It is a special item if it actually *does* contain an echo canceller. Ops, I don't use mux's but I know the echo can function has been around for years in some telco grade mux's. Since I've not shopped for any in the last several years, didn't realize they were optional 'features' for some vendors. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Zaptel dacs / dacs
On Fri, Jul 02, 2004 at 02:48:17PM -0700, Chris A. Icide wrote: 2) can you cross connect PRI interfaces? in other words can you use the dacs functionality to insert a digium card (on a system running asterisk) in between a pri from a carrier, to a legacy pbx system? Yes, you can. But what's the point ? With such a setup (zaptel bridging), you don't get that many benefits. I developed a small std-local driver that can help you : - cross connect two PRIs - and get a monitoring feed (imagine a Y cable), so that you can do (read-only) analysis on the PRI traffic. But I doubt that's what you want to do there. What you may want to do is being able to use the legacy PBX, as well as Asterisk features. For that, you need to : - setup Asterisk for the PRI connection to the telco (pri_cpe) - setup Asterisk for the PRI connection to the legacy PBX (pri_net) - setup your dialplan (extensions.conf) so that : - Asterisk talks to the telco the right way - Asterisk talks to the PBX the right way - Asterisk forwards calls to/from the PBX the right way - Asterisk does whatever else you want it to do It's not that hard. However, it will probably involve some trial and error, and your PBX users might not like it. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Asterisk bounty: SIP simultaneous registry
http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Multiple E1s over TDMoE?
On Sat, Jul 03, 2004 at 07:44:54PM +0200, Thilo Salmon wrote:
How would you go about running, 8 or 16 say, E1s over TDMoE? Would you
create multiple dynamic spans or just one large one? How would you
assign d channels to spans, if you had just one large span?
Did any of you guys try this before?
Somewhat, yes.
I run 4 TDMoE E1s between pairs of servers (4 E1s between each of
them).
What I think about that :
- it works
- TDMoE doesn't like SMP. It doesn't like running on a NIC used for
other kind of traffic. It will crash your box under heavy (non
TDMoE) load. I believe that there must be some race condition
related to dev_queue_xmit(), which is probably not callable at
anytime.
- the subaddr support is not complete in the released driver. Here is
a patch that will handle it (as described in zaptel.conf)
- it is not 100% reliable. You will get frame drops, and you will
notice it if you look at your D-channel dumps.
- use high quality NICs and switches.
--
Nicolas Bougues
Axialys Interactive
--- ztd-eth.c.old 2004-02-01 06:53:58.0 +0100
+++ ztd-eth.c 2004-07-11 00:51:45.0 +0200
@@ -251,7 +251,7 @@
{
struct ztdeth *z;
char src[256];
- char tmp[256], *tmp2, *tmp3;
+ char tmp[256], *tmp2, *tmp3, *tmp4 = NULL;
int res,x;
unsigned long flags;
@@ -273,6 +273,7 @@
return NULL;
}
if (tmp2) {
+ tmp4 = strchr(tmp2+1, '/') +1;
/* We don't have SSCANF :( Gotta do this the hard way */
tmp3 = strchr(tmp2, ':');
for (x=0;x6;x++) {
@@ -288,7 +289,8 @@
} else
break;
if ((tmp2 = tmp3))
- tmp3 = strchr(tmp2, ':');
+ if (!(tmp3 = strchr (tmp2, ':')))
+ tmp3 = strchr (tmp2, '/');
}
if (x != 6) {
printk(TDMoE: Invalid MAC address in: %s\n, addr);
@@ -300,6 +302,25 @@
kfree(z);
return NULL;
}
+ if (tmp4) {
+ int sub = 0;
+ int mul = 1;
+
+ // We have a subaddr
+ tmp3 = tmp4 + strlen (tmp4) - 1;
+ while (tmp3 = tmp4) {
+ if (*tmp3 = '0' *tmp3 = '9') {
+ sub += (*tmp3 - '0') * mul;
+ } else {
+ printk(TDMoE: Invalid subaddress\n);
+ kfree(z);
+ return NULL;
+ }
+ mul *= 10;
+ tmp3--;
+ }
+ z-subaddr = htons(sub);
+ }
z-dev = dev_get_by_name(z-ethdev);
if (!z-dev) {
printk(TDMoE: Invalid device '%s'\n, z-ethdev);
@@ -311,7 +332,7 @@
for (x=0;x5;x++)
sprintf(src + strlen(src), %02x:, z-dev-dev_addr[x]);
sprintf(src + strlen(src), %02x, z-dev-dev_addr[5]);
- printk(TDMoE: Added new interface for %s at %s (addr=%s, src=%s)\n,
span-name, z-dev-name, addr, src);
+ printk(TDMoE: Added new interface for %s at %s (addr=%s, src=%s,
subaddr(net byte order)=%d)\n, span-name, z-dev-name, addr, src, z-subaddr);
spin_lock_irqsave(zlock, flags);
z-next = zdevs;
@@ -350,3 +371,6 @@
module_init(ztdeth_init);
module_exit(ztdeth_exit);
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry
http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry Updated, Allow a SIP device to register more than once so a single extension may exist in multiple locations. Upped total to $75. Daniel... Daniel Jimenez wrote: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] T1 Hardware Echo Can
Hey Brent, If you only need a small number of fxo ports, pstn gateways in the form of Cisco boxes, Mediatrix 1204, as well as many other products can do that at a relatively inexpensive cost. The echo can function is built into those boxes, which do work. Each comes with additional considerations or issues however. The mediatrix boxes do look good. I'll have to consider for next project. There are a couple of things to watch out for on the Mediatrix that aren't well published. - their only support is through their resellers, and most of those seem to be traditional pbx folks that don't have a background involving asterisk - they are highly focused towards 1104 to 1204 compatibility and toll bypass, and have little interest in ensuring rfc compliance or support for non-Mediatrix interfaces - all software upgrades/fixes are chargable - silly little things get in the way (eg, better have the same ring cadance on all four pstn lines attached to the 1204, or the 1204 won't answer some lines) - excellent echo can functions, worked like a charm after spending a very large amount of time tweeking poorly documented technical parameters through a PC SNMP (only) management interface. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] T1 Hardware Echo Can
If you install a T1 card and an external T1 mux (with fxo cards), the echo can function already exists within the mux and/or cards. Don't really need 'another' external echo can box unless you actually purchased a T1 mux that didn't have echo can in the first place (and they do exist). Just to be clear here there are a lot of channel banks, fact most that are suggested here (ta 750, ab i/ii, adit 600 and zhone etc) that dont have have echo cans that do benefit from the hardware echo cans eg Since there has been more then a few people interested in fxo channel banks, do you (or anyone else) have any recommendations for specific models that have optional echo cans? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] VoIP hackers gut Caller ID
On Wed, Jul 07, 2004 at 11:57:31AM -0400, Timothy R. McKee wrote: This has always been one of my pet peeves, even as I worked in the industry. A telco switch operating a DS1 on trunk side should enforce caller-id numbers to be within the range of DID numbers assigned to that trunk. There should be a default DID number that is used to replace any *invalid* numbers sent on that trunk. Note that blocked caller ids would still be blocked, but the rest of the data should be corrected. Blocking ID is ok, lying about it is not. Blind trust of a non-SS7 link is a _bad_ thing. PRI signalling enables Network provided or User provided caller-id. Maybe IAX could implement such a thing. It's very common in France (at least) : - the network will provided a guaranteed caller-id - the user (CPE) may provide another one (usually, a DID number) and the called party gets both. Unfortunatly, as far as I understand, Asterisk is not really designed to handle more than one caller id number. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NuFone Error
Use the NuFone to call numbers 1800,1866 and after 20 seconds the call isinterrupted. In log of * it writes: Max retries exceeded you host. Somebody can help me? Thank you Joao Carlos Moura
Re: [Asterisk-Users] NuFone Error
You need to send a vallid CALLERID to Nufone. On Sat, 10 Jul 2004, V59Net wrote: Use the NuFone to call numbers 1800,1866 and after 20 seconds the call is interrupted. In log of * it writes: Max retries exceeded you host. Somebody can help me? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] NuFone Error
No you don't it will just make one up... bkw - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 10, 2004 8:17 PM Subject: Re: [Asterisk-Users] NuFone Error You need to send a vallid CALLERID to Nufone. On Sat, 10 Jul 2004, V59Net wrote: Use the NuFone to call numbers 1800,1866 and after 20 seconds the call is interrupted. In log of * it writes: Max retries exceeded you host. Somebody can help me? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for a patch that was post May 1 2004
Hello group I'm working on getting festival installed and working on my FC1. I ran into a problem and after searching Google I found this message talking about a patch for Speech Tools and Festival http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html The above site does not have the files. Does anyone in the group have this patch? Marc Sutter Reed Wade do you still have it? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two server
Hello
Use two Asterisk servers.
I registered Server2 in Server1. When I bind for an extension in the
Server1, the hard call some as and is interrupted.
My configuration:
iax.conf / Server 1
[20001]
type=friend
accountcode=20001
host=dynamic
secret=secret
context=sip
disallow=all
allow=gsm
auth=plaintext
//
iax.conf / Server 2
register = 20001:[EMAIL PROTECTED]
[20001]
type=user
context=fromiax
auth=plaintext
Extension / Server2
exten = _33.,1,SetCallerId,${CALLERIDNUM}
exten = _33.,2,Dial(IAX2/20001:[EMAIL PROTECTED]/${EXTEN:2})
exten = _33.,3,Congestion
Error
-- Executing Dial([EMAIL PROTECTED]/17,
IAX2/20001:[EMAIL PROTECTED]/123456) in new stack
-- Called 20001:[EMAIL PROTECTED]/123456
-- Call accepted by 24.232.89.22 (format GSM)
-- Format for call is GSM
-- Operating with different codecs, can't native bridge...
-- Hungup
== Spawn extension (sip, 33123456, 2) exited non-zero on
'[EMAIL PROTECTED]/17'
-- Hungup '[EMAIL PROTECTED]/17'
Necessary of aid. It forgives my English.
Thanks
Tiago
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Re: [Asterisk-Users] Looking for a patch that was post May 1 2004
Look in your /usr/src/asterisk/contrib directory Hello group I'm working on getting festival installed and working on my FC1. I ran into a problem and after searching Google I found this message talking about a patch for Speech Tools and Festival http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html The above site does not have the files. Does anyone in the group have this patch? Marc Sutter Reed Wade do you still have it? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] NuFone Error
On Sat, 10 Jul 2004, Brian K. West wrote: No you don't it will just make one up... I beg to differ, Mr Brian sir. I had problems calling 800 numbers with Nufone, and Jeremy explained to me that they check for a caller id before sending calls to 800 numbers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP provider for 2 site enterprise deployment??
Hi All, Looking for a VoIP provider for a 2 site IP-PBX deployment to provide DID numbers for each person in the offices (~75 numbers across the 2 sites) and outbound VoIP calling. The sites will have sufficient POTS line for backup outbound and 'main number' inbound calling. I haven't gotten anywhere with VoicePulse on something different than their connect service or with nufone. Any provider suggestions or links much appreciated. If you are a provider that could handle this kind of configuration please send me an email with info. Thanks, Jim jobrien A-T bridgeport-networks D-O-T com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Avoiding transcoding
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Pajari Sent: Saturday, July 10, 2004 4:58 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Avoiding transcoding Dr. Murphey: How can one specify that codec selection should avoid transcoding if possible? Asterisk only transcodes if the original audio stream needs to be connected either to (a) another audio stream or (b) an internal Asterisk function -- and the original audio stream is encoded in a format that is not compatible with the other audio stream or Asterisk function. The reason I ask is that when enum lookup succeeds and the destination only accepts ULAW, the various transcodings seems to garble the audio if GSM, ILBC, etc. are allowed. To illustrate the point above further, if you have a destination that only accepts ULAW, and an audio stream in GSM or ILBC, you must transcode. It is simply not an option since ULAW is not GSM, Asterisk has to convert (transcode) the ULAW samples into GSM samples and vice versa. Is it possible to configure asterisk to select a codec common to both channels? Per your example, I tested a call from XTEN (sip) that allowed ULAW or GSM, through asterisk, to a destination that accepts only ULAW. XTEN(SIP) - asterisk - SIP destination It should be possible to avoid transcoding, but asterisk shows this: Peer User/ANRCall ID Seq (Tx/Rx) Format 216.234.116.184 1800246846 1b07a93b13f 00102/0 ULAW 172.16.0.100 xtenEEE6B555-5B 00101/09156 GSM Because the destination is found by enum lookup, it's not possible to know which protocol or codec in advance. Thank very much for your explanations. Cheers, Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Three (quick?) questions...
Hi, T1 is the carrier. T1 provides 24 D channels of 64Kbps each. Telephone companies provide ISDN (integrated services data network) on top of T-carrier. Two common flavors are BRI (basic rate interface) and PRI (Primary rate interface.) BRI provides two 64 kps channels, PRI provides 23 usable channels, the 24th is used for signalling. So--you can get phone calls over a T1 or over a T1 that is provisioned as a PRI. You can get 24 calls on a T1 and 23 on a PRI. A T1 has 24 channels. You can split, that is partialize, the channels between data and voice. You can do this with hardware outside the * server. Higher end Cisco routers, for example, support this. You can also use * and linux to partialize the T1. You better plan on spending a lot of time on making it work if you do it this way. You have to install the Linux packages to split the line. NON trival. Works great, though. Much less expensive, too. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Saturday, July 10, 2004 8:33 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Three (quick?) questions... [Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones inside? [In other words, I obviously need a NIC -- but what would be the bare-bones telco POTS interface?] 2) What phones would be recommended for inexpensive (doesn't even need LCD), and yet functional? 3) In order to share data and voice over a T1, does it have to be PRI? [I've got a T1 I could probably play with, but I'd like to be sure it'll... well, you know: work.] Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] New Asterisk bounty: SIP simultaneous registry
I'm not sure I understand what you are trying to do. You have an administrative assistant and several other staff. You want the administrator to be able to take calls directed to the staff extensions? If I have the requirement right, you could accomplish this by ringing the staff extension and the admin extension at the same time. The Dial command allows you to ring multiple extensions simultaneously. If you want to be able to more easily recognize what extension the traffic if for, you can add additional extensions to the 7960. For example, if you have two staff the admin monitors, add two additional extensions to the 7960. The admin can tell who is being called by the extension that rings. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Jimenez Sent: Saturday, July 10, 2004 3:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+s imultaneous+registry Updated, Allow a SIP device to register more than once so a single extension may exist in multiple locations. Upped total to $75. Daniel... Daniel Jimenez wrote: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultane ous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Three (quick?) questions...
Hi Paul, you would know better than I would but I always thought a T1 was 24 channels of voice with the signalling additional like we have in Australia a Pri or E1 is 30 channels voice channels plus signalling. Can anyone else clarify? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Sunday, 11 July 2004 2:39 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Three (quick?) questions... Hi, T1 is the carrier. T1 provides 24 D channels of 64Kbps each. Telephone companies provide ISDN (integrated services data network) on top of T-carrier. Two common flavors are BRI (basic rate interface) and PRI (Primary rate interface.) BRI provides two 64 kps channels, PRI provides 23 usable channels, the 24th is used for signalling. So--you can get phone calls over a T1 or over a T1 that is provisioned as a PRI. You can get 24 calls on a T1 and 23 on a PRI. A T1 has 24 channels. You can split, that is partialize, the channels between data and voice. You can do this with hardware outside the * server. Higher end Cisco routers, for example, support this. You can also use * and linux to partialize the T1. You better plan on spending a lot of time on making it work if you do it this way. You have to install the Linux packages to split the line. NON trival. Works great, though. Much less expensive, too. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Saturday, July 10, 2004 8:33 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Three (quick?) questions... [Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones inside? [In other words, I obviously need a NIC -- but what would be the bare-bones telco POTS interface?] 2) What phones would be recommended for inexpensive (doesn't even need LCD), and yet functional? 3) In order to share data and voice over a T1, does it have to be PRI? [I've got a T1 I could probably play with, but I'd like to be sure it'll... well, you know: work.] Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] New Asterisk bounty: SIP simultaneous registry
Paul, The question is very simple. When I call a SIP user, the phone should ring in more than one extentions. Also more than one phone should be able to register with asterisk. Right now it is not the case. The last phone which register will be receiving the calls. This type of situations might be needed in call centres. Called 12345 |---(12345) Ringing |---(12345) Ringing |---(12345) Ringing So you don't need to disturb asterisk when you add more devices to it to receive calls. Such facility is not available in asterisk at this moment. I hope this helps. Since I feel this is a great feature, I will topup up to $100/- -.Kannaiyan http://www.goods2world.com -- Your Only VoIP - Original Message - From: Paul Mahler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 5:44 AM Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry I'm not sure I understand what you are trying to do. You have an administrative assistant and several other staff. You want the administrator to be able to take calls directed to the staff extensions? If I have the requirement right, you could accomplish this by ringing the staff extension and the admin extension at the same time. The Dial command allows you to ring multiple extensions simultaneously. If you want to be able to more easily recognize what extension the traffic if for, you can add additional extensions to the 7960. For example, if you have two staff the admin monitors, add two additional extensions to the 7960. The admin can tell who is being called by the extension that rings. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Jimenez Sent: Saturday, July 10, 2004 3:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+s imultaneous+registry Updated, Allow a SIP device to register more than once so a single extension may exist in multiple locations. Upped total to $75. Daniel... Daniel Jimenez wrote: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultane ous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] New Asterisk bounty: SIP simultaneous registry
On Sun, 11 Jul 2004, Kannaiyan Natesan wrote: When I call a SIP user, the phone should ring in more than one extentions. Also more than one phone should be able to register with asterisk. Right now it is not the case. The last phone which register will be receiving the calls. This type of situations might be needed in call centres. I think I understand now what you're looking for. But under an arrangement like this, how will asterisk know when a phone which had registered from some IP has re-registered itself sometime later on a different IP? Such a situation could happen in a dhcp environment. Automatic time-outs may be able to avoid or minimize the impact of something like this. What other difficulties might come up? Although the idea does have appeal, it seems like the increased potential for problems outweighs any inconvenience incurred by modifying a line in extensions.conf. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] New Asterisk bounty: SIP simultaneous registry
in other SIP proxy server, this can be done easily, i mean its default 1 or more phone could be registered at 1 number (12345) and resulting same effect as u ask SER (SIP Express Router, http://iptel.org/ser) can deal with this SER is a friend to asterisk, i think :), you can accept calls with SER and pass it to asterisk to process complex dialplan but if this feature implemented in asterisk alone, it would be nice *** REPLY SEPARATOR *** On 11/07/2004 at 6:00 Kannaiyan Natesan wrote: Paul, The question is very simple. When I call a SIP user, the phone should ring in more than one extentions. Also more than one phone should be able to register with asterisk. Right now it is not the case. The last phone which register will be receiving the calls. This type of situations might be needed in call centres. Called 12345 |---(12345) Ringing |---(12345) Ringing |---(12345) Ringing So you don't need to disturb asterisk when you add more devices to it to receive calls. Such facility is not available in asterisk at this moment. I hope this helps. Since I feel this is a great feature, I will topup up to $100/- -.Kannaiyan http://www.goods2world.com -- Your Only VoIP - Original Message - From: Paul Mahler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 5:44 AM Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry I'm not sure I understand what you are trying to do. You have an administrative assistant and several other staff. You want the administrator to be able to take calls directed to the staff extensions? If I have the requirement right, you could accomplish this by ringing the staff extension and the admin extension at the same time. The Dial command allows you to ring multiple extensions simultaneously. If you want to be able to more easily recognize what extension the traffic if for, you can add additional extensions to the 7960. For example, if you have two staff the admin monitors, add two additional extensions to the 7960. The admin can tell who is being called by the extension that rings. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Jimenez Sent: Saturday, July 10, 2004 3:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+s imultaneous+registry Updated, Allow a SIP device to register more than once so a single extension may exist in multiple locations. Upped total to $75. Daniel... Daniel Jimenez wrote: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultane ous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://sleepless.ngoprek.org VoIP Rakyat: (0921) 20006 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] VoIP hackers gut Caller ID
Hi Guys, This topic has become pretty much pointless. CallerID was never designed to be any kind of authentication scheme. Also, it is very hard for telco to restrict proper usage of CallerID in PRI or SS7 (Please consider number protability, etc.) We all already agreed on fact that author of this article are moron. Let's not discuss any ideas of making CallerID secure or ajusting IAX to carry 2 or 3 CallerID records. All of this is pointless. If someone conducts business based on CallerId, it's up to them. If somebody comits crime with fake CallerID, it's also fine. People, this world is not perfect. There are thousands of telco companies where you will be able to find somebody who does not enforce proper CallerID. There are bunch of telephony guys who can do a lot of stuff, which you can't even think about it. But people, please do not write articles like that and do not publish it on MSNBC, NY Times and CNN. Thanks, Aleksandr Palatkevich BPVN Technologies Inc. http://www.pipeboost.com/ Phone: (917) 723-0306 Fax: (212) 937-2170 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Bougues Sent: Saturday, July 10, 2004 7:34 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID On Wed, Jul 07, 2004 at 11:57:31AM -0400, Timothy R. McKee wrote: This has always been one of my pet peeves, even as I worked in the industry. A telco switch operating a DS1 on trunk side should enforce caller-id numbers to be within the range of DID numbers assigned to that trunk. There should be a default DID number that is used to replace any *invalid* numbers sent on that trunk. Note that blocked caller ids would still be blocked, but the rest of the data should be corrected. Blocking ID is ok, lying about it is not. Blind trust of a non-SS7 link is a _bad_ thing. PRI signalling enables Network provided or User provided caller-id. Maybe IAX could implement such a thing. It's very common in France (at least) : - the network will provided a guaranteed caller-id - the user (CPE) may provide another one (usually, a DID number) and the called party gets both. Unfortunatly, as far as I understand, Asterisk is not really designed to handle more than one caller id number. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
