[Asterisk-Users] Using US Robotic router for 60 calls
Hi, In order to save public IPs, I am attempting to use a Router SureConnect of US Robotics in order to route calls to Asterisk on a private IP. Would you recommand a large router like Cisco if we have 30 calls or a normal router can do ? Any advise is greatly appreciated ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceBlue GSM
Hi, > > That is interesting. What is the make and the model that you are > > referring to? Is there a website with more info? As for the models, we sell them as OEM. You may contact me offlist if interested. Better priced and more powerful than existing devises out there. > > > > I currently use Quescom IP400 GSM but they are expensive (although they > > support up to 12 GSM channels). > > Quescomm, are good but have drawback. 12 SIMs model you need to unscrew to change the SIMs. This is not very practical. I recommand other brands like Vierling, who are more practical in everyday life of a corporate grade carrier devise. > > Can these GSM modules work as proxies so when you are local your GSM > handset can be used on the VoIP network? When you walk away from the > nanocell/picocell transmitter it autoswitches to the real provider. > The idea is brilliant. I haven't seen such product on market. Tellular have something similar, but you still need some workout. I have other solution for this functionality: In europe you can have two SIMs for same phone number. In Belgium for instance (proxi-Duo). Then, you plug one SIM in the GSM Gateway and another one in the GSM devise. When you get home you switch your GSM and automatically the other sIM becomes active, when you walk away you switch your GSM phone on and it will automatically become active. > The way GSM auth works this technically would not be that difficult to > do, and infact there is equipment that does not interface to anything > that does proxy call setup/tear down info (IMSI catchers for example), > so what I am asking for is not that far out. Some GSM network operators do this. Last SIM registered is one to be considered and others become inactive. > > This would also make it easier for call shops, set up one of these and > people can use their mobile as a voip phone. Hmm. What is added value of customers using their cell phone in callshops? They have walked all the way long to reach the place. You may explain further In the office your GSM > calls are sent to asterisk to your desk phone and/or your mobile, > outbound calls go over asterisk for least cost routing, etc. But the > handset is your mobile (and in theory on your person at all times). > > If anyone knows of a device that integrates to asterisk that does that I > would *greatly* appreciate hearing from you regarding a vendor, > make/model, even a supplier. If you are a supplier I grant you > permission to use my contact info to directly contact me about that > issue only so long as you dont add me to any lists. > Other idea is that poeple > Thanks > > -- > Trixter http://www.0xdecafbad.com > UK +44 870 340 4605 Germany +49 801 777 555 3402 > US +1 360 207 0479 or +1 516 687 5200 > FreeWorldDialup: 635378 > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceBlue GSM
Etienne, I am not sure I understand all what you require. Do you need to know the cost of the voiceblue of 2N or you need to find solution that can allow you send GSM calls ? There are several alternatives: 1-) Voiceblue as you mentioned; 2-) You can buy a voip2GSM Gateway. To which you no longer need hardware, you just register the voip2GSM devise to Asterisk and then it is ready to receive and send calls just like any other sip phone. Cost of this is around 400 USD / UNIT When you talk about sms capability, dyou want to originate or receive SMSs through the devise? Selon Etienne Pretorius <[EMAIL PROTECTED]>: > Hello All * users. > > I have been looking for a way to allow GSM termination through Asterisk > to occur. I am not a Telekom fundi and I have set up IAX2/IAX/SIP on > asterisk with the ZAP channels via the Digium TDM 400P. I am unable to > find any place that can tell me the cost of the VoiceBlue with a > currency to I can calculate the cost of buying one. Alternativly - or > just out of interist - I only really need to handle one GSM call @ a > time and have a SMS capability... is there anyone that can suggest the > best way to do so without doing a hack/patch to make a device to > interact with asterisk? > > -- > Kind Regards > Etienne > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank of E1s? (one E1 input --> 2 x E1 output)
I just called this company. They seem to do what is required. Now remains the pricing part of it. I will wait for their feedback. http://www.megatelindustries.com/products.htm Hakem, Selon Julio Arruda <[EMAIL PROTECTED]>: > Matteo Brancaleoni wrote: > > yes, some multiplexer allows that, but they're quite expensive > > compared to another E1 card for asterisk. > > I think you'll need at least 1k $$$ for a such splitter. > > > > > Matteo, would you have any reference for this 'mux/splitter' ? > I would guess it need to be smart enough to dig into the signalling, > since is not only the PCM DS0s that would need to be "Y-splitted". > [], > > > > > Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto: > > > >>Hi, > >> > >>Assume I have one E1 digium card to which I want to plug two distinct E1 > PABXs , > >>one with 15 channels and the other with 15 channels; > >> > >>Is there a sort of E1 multiplexer devise that allows me to plug in one hand > the > >>E1 port of the Digium card and on the other hand the two PABXs? In this > same > >>devise, I should be able to say that 15 channels need to go to first > Interface > >>and 15 other channels need to go to other interface. > >> > >>Or is it necessary to acquire a another E1 card although I don't need to > process > >>more channels (30 channels are ok). > >> > >>Any help is greatly appreciated. > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel bank of E1s? (one E1 input --> 2 x E1 output)
Hi, Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs , one with 15 channels and the other with 15 channels; Is there a sort of E1 multiplexer devise that allows me to plug in one hand the E1 port of the Digium card and on the other hand the two PABXs? In this same devise, I should be able to say that 15 channels need to go to first Interface and 15 other channels need to go to other interface. Or is it necessary to acquire a another E1 card although I don't need to process more channels (30 channels are ok). Any help is greatly appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Shanghai or Bangalore DIDs
Can Offer Hong Kong, 10 USD/DID/month . Set-up cost 20 USD. Minimum committment is 06 months. Regards, Hakem, Selon Marc Storck <[EMAIL PROTECTED]>: > I'm also looking for numbers from > > HongKong, > Taiwan, > Japan and > Singapore > > So if someone has some DIDs from this areas, I'm very interested to get > one or another from those DIDs. > > Best Regards, > > Marc > > Marc Storck wrote: > > Hello, > > > > does someone offer DIDs from the areas of shanghai and/or bangalore. > > > > Many thanks, > > > > Marc > > > > > > > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > CTOMarc Storck > MS Networks SA [EMAIL PROTECTED] > IT Service Providerhttp://www.msnetworks.lu > 15, route d'Esch Phone: +352 2727 3030 > L-4450 Belvaux Fax: +352 2727 3060 > > --- MS Networks powered service --- > http://www.LuxAdmin.com Hosting and housing solutions > --- > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards
Thanks very much for this info Andrew. Selon Andrew Kohlsmith <[EMAIL PROTECTED]>: > On April 25, 2005 07:39 am, [EMAIL PROTECTED] wrote: > > I would like to setup a 8 FXS + 8 FXO interface using the X100P Cards. The > > problem now comes in the PCI ports. Is there any PC that can handle 16 > > ports? > > > > What is most optimal solution? > > The most optimal solution would be a TE110P + a channel bank. The TE110P is > about US$500 and a channel bank with 8FXS and 8FXO (with option to expand to > 8 more ports) will run probably US$700-1000 on ebay. > > There is 1 PCI card in your computer and a piece of external equipment (the > channel bank). You could go with 4 TDM400Ps to get the same number of ports > but you will undoubtedly have trouble with sharing IRQs and the interrupt > overhead is going to eat you alive. > > Channel banks are great; the better ones (Adit600) can do far-end disconnect > supervision and I think pretty much all of them do dynamic impedance > adjustment, meaning they're FAR less prone to echo. Just about anyone's FXS > modules work, but be careful with FXO modules on channel banks. Access Bank > I and IIs do *NOT* do far-end disconnect, meaning if someone on the other > side hangs up, Asterisk won't be able to tell. > > -A. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards
Hi, I would like to setup a 8 FXS + 8 FXO interface using the X100P Cards. The problem now comes in the PCI ports. Is there any PC that can handle 16 ports? What is most optimal solution? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 not work with DTMF and AGI
Hello, I am stuck while attempting to insert DTMF commands from a SIP gateway to an IVR menu running AGI php scrit. If I do: SIP phone --> ulaw --> IVR then dtmf works fine If I do: SIp gateway --> g729 --> IVR then the mneu still works but does not accept DTMF ? I have tried to set dmtfrelax=yes and dtmfmode=info, also tried dtmfinfo=rfc2833 Still did not work, any clues please? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF does not work with g729 and AGI
Hello, I have an AGI script that runs a menu at two levels of a tree. If I call the extension from a voip phone with g711, the menu works fine and accepts DTMF no probs. Then, when I Call from a DID, it sends call using SIP and g729 to¨* box. The IVR also starts running, but no DTMF is deteced. I have tried various configs (combinations of dtmfmode=info, dtmfmode=rfc2833 and dtmfrelax=yes, dtmfrelax=no) with no success. Any hint? sip.conf [SS_SIP] type=peer host=XXX.XX.XXX.XX dtmfrelax=no ;dtmfmode=rfc2833 dtmfmode=info context=outbound disallow=all allow=g723.1 allow=g729 [SS_SIP] type=user host=XXX.XX.XXX.XX context=outbound dtmfmode=inband disallow=all allow=g723.1 allow=g729 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please make sure there is subject in mails
Selon Rich Adamson <[EMAIL PROTECTED]>: > > > > Funny, they sell these old cards.. it seems like they are selling > refurbs > > > > as new.. ... anyways RMA is on its way, would be nice if they would > send > > > > one as a replacement first, so that we could continue our work and > don't > > > > have to delay it. > > > > > > They can, its called cross-shipment, but they need a credit card number > > > to ensure they get your return shipment. You have to ask for it. That's > > > the way I did it. > > > > Thanks for the info .. > > > > > Regarding the refurbs, if you or I were owners of digium, how would we > > > handle a backstock of older (possibly refurb) cards when its somewhat > > > known the old cards work fine in some systems? (And, we don't have a > > > clue which systems/motherboards the cards worked fine in.) > > > > Well I guess one can also add location, as each phone provider will have > > different class 5 switches one connects to with different signaling > > basically .. But still, they should fix the problem once and for all with > > allowing flash firmware upgrades... hell even Adaptec can manage that .. > > For the TDM card its not a flash firmware issue. Based on what others > have reported, etc, the Rev E/F card had a missing trace on the circuit > board (others observed an added jumper a month or so after the card came > out), the Rev H card (and maybe other rev's before that) had added > components on the circuit board, plus the fxo modules apparently changed > since the originals came out. The added jumper had something to do with > module slot 1. > > Personally, it wouldn't bother me a bit if I received a replacement card > with an added jumper as long as the card worked as expected. > > So, digium probably has a back-stock of earlier rev levels that might > work just fine, but adding a jumper would not change the Rev level > reported to the system. Without knowing specifically what was changed > on each Rev, there is no way to guess at how refurb'ed cards should > be handled. I hope the current Rev is stable, etc. > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best FXO Voip Gateway for Asterisk
Quintum are good Selon Chad Brown <[EMAIL PROTECTED]>: > There are many analogue gateways to choose from: > http://www.voip-info.org/wiki-VoIP+Gateways > > Does anyone have experience with several that could point me in the > right direction? I need 5-8 ports. At some point I see us going digital > but I'm not sure when TCO will make sense. > > Advice based on real world experience would be much appreciated. > > Thanks, > > Chad > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] Re: International callback strategies
Then, I realised a spent lot of time thinkin about this solution. Other option is that you put a prepaid calling card platform in Russia. I saw in CEBIT some russian companies selling prepaid calling cards. In order to give access to your customers without them to know where is the platform, you can also sell them "dialers". Dialers call platform and hence people won't know where it is located. Maybe this goes beyond scope but it is workth knowing such solutions exist Selon Adam Goryachev <[EMAIL PROTECTED]>: > On Sun, 2005-04-10 at 15:39 -0700, snacktime wrote: > > On Apr 10, 2005 3:17 PM, Hakem Taourchi <[EMAIL PROTECTED]> wrote: > > > > 2-) You can create DID system. That is, you buy 1000 DID, and each > > > customer has got a dedicated US did. So when your Callback Systems > > > receive call on DID 101 (without hang up), the callback system knows > > > upfront who to callback; > > > > > This seems to be the solution that will work the best, although for a > > small overhead for the did's. > > Can you get russian DID's routed to you via VoIP?? Then your russian > users call a local number to get your system, and then can use disa or > astcc or something to make the destination call to USA?? > > PS, unless of course you can't get DID locally in russia, then maybe you > can setup small PC with one/two analog lines and decent internet in > someone's home you know (mother/brother/something) ?? > > Regards, > Adam > > -- > -- > Adam Goryachev > Website Managers > Ph: +61 2 8304 [EMAIL PROTECTED] > Fax: +61 2 9345 4396www.websitemanagers.com.au > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk based Call Accounting software- 1strelease
If you will donate some money to AreskiCC prepaid platform developer, he may want to extend module to pure accounting. I will also donate some money to a linux - web based accounting system Selon "Chris Mason (Lists)" <[EMAIL PROTECTED]>: > > There are other accountings available, and a Windows solution is for ME > > the very last choice!!! > > Really? I am about to commission someone to write a call accounting package > for us because there is nothing out there. Exactly what else is available? > > Chris Mason > www.anguillaguide.com > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk based Call Accounting software - 1strelease
Ronald, What are other solutions available that can offer web based access and work with Asterisk? Selon Ronald Wiplinger <[EMAIL PROTECTED]>: > Chris Mason (Lists) wrote: > > >Call Accounting is such an important issue for me it is literally a make or > >break component, without it I will not be able to deploy Asterisk at our > >resort. If I have to use a windows computer to download and run the client > >end of the software, so be it. At least the software will work and I will > >have a solution. I think you should be more appreciative they are > >accommodating Asterisk and less dogmatic about platform issues. > > > > > > > Sure, it is your choice! > You can use whatever is good for you. > > For me it is not possible, ... and I will not use it, ... that is MY > choice too. > > Unfortunately there are not even screenshoots, so that you could imagine > what it could do! > > There are other accountings available, and a Windows solution is for ME > the very last choice!!! > > > bye > > Ronald > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank replacement
Maybe following options: 1-) Get another channel bank from ebay at low cost. Which will also need another T1 card; 2-) Use 40 voip phones at 50 USD each and you no longer need the card neither the channel bank. But a reliable local network ; Selon Peter Hoppe <[EMAIL PROTECTED]>: > Hello, > > I am working for a charity in the UK and I am projecting a new phone system. > > We would like to connect our two-wire telephones (40 or so) to an ADIT > 600 channel bank, and connect that into an Asterisk box via the CMG card > or T1 card. > > I have been in talks with Carrier Access about the purchase of a new > channel bank and we tried to get a minor version of it first for testing > with the intention of upgrading to the full product if we are happy with it. > > Unfortunately since a few months I cannot get any further with CAC, as > they keep not coming back to us on how we proceed. I feel that the > channel bank would be the best solution, but it seems that we are just > to small fish to fry for them. > > So - would there be any other way to connect 40+ telephones (two wire) > into an asterisk box? Are there any voip gateways that actually conform > to SIP standard (unlike what I heard from the Mediatrix voip gateways > 1124 and 1204 which seem to use non standard SIP and have > pay-as-you-upgrade)? > > Thank you very much for your consideration! > > Peter Hoppe > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Sangoma VS. Digium
It seems like Digium is good for beginners and Sangoma is good for experts in a nutshell. What I need to know is whether I can offer Internet Dial-up Access using Sangoma hardware or Digium hardware? This is because pricing per E1 is attractive compared to existing proprietary solutions Has anyone done it? Thanks, Selon Scott Stingel <[EMAIL PROTECTED]>: > Good question. I've had good luck with the Digium TE405P recently in a > multiple T1 install. > > Don't let the discussion scare you too much! > > -Scott Stingel > > > cmould wrote: > > > Where is this discussion going. I am about to do an installation that > > will require t1 interfaces. I am new to the telephone world and found > > the original discussion useful. > > > > I need to know from a reliability and performance standpoint what is > > the better choice. Sangoma or Digium? > > > > > > > >___ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] open source Asterisk Application of the year?
Hello, I was just wondering if there were a prize like the open source application of the year relative to Asterisk? All these developer doing good job and all free need some present sometime that we can all donate. Anything like that exists? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM Hardware Setup
Do you want to make GSM calls? Or just use PCMCA card for the data connection? Inb case you want to make GSM calls with SIM cards, you need to get GSM Gateways or called sometimes fixed-cellular terminals and plug them to a FXO card. Or you can get a voip->GSM box. Selon Jan Kellerhoff <[EMAIL PROTECTED]>: > Hello folks, > > I am a newbie, having a little question, about GSM-Asterisk Hardware. > I'd like to set up a small PBX for GSM use, and ask myself it is possible > to use any GSM Card phone (PCMCIA) over some PCMCIA to PCI controller? > > Would be easyer than the cellphone socket, as I wouldn't need to buy new > Hardware. > Maybe someone knows...? > > > Kelly > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma VS. Digium
Yes, Most hardware manufacturers I know sell directly at retail price. In voip Business and from my experience, you can order Quintum gateways from Quintum Technologies right away at retail price. You can always get them cheaper from reseller. GSM devices manufacturers sell direct as well although they have resellers, unless exclusively agreed not to sell in that part of the world. For me it seems logical that manufacturer promote products themselves and sell them. Because resellers sometimes just don't react to sales leads. There is no reason why a manufacturer should lose these sales leads if reseller does not react. Now, if customers go to Digium right away instead of coming to reseller even though reseller is better price and close with pieces in stock to be served immediately, I think it is not the distribution of Digium that has problem, but the customers :-) Regards, Hakem, Selon Matteo Brancaleoni <[EMAIL PROTECTED]>: > Hi, > > Il giorno gio, 07-04-2005 alle 13:02 +0200, [EMAIL PROTECTED] ha scritto: > > > Digium do sell online and so many other of their resellers do. The > important > > point is that they don't sell lower cost than their resellers, which is the > > case. > Please find an hardware producer that sells directly to endusers, when > they have also distributors/resellers. > The way is: if you have resellers, sell through them. if not directly to > end user. > > > Reseller added value is find customers and retail locally in his place with > > local variables of config, ...etc. They are the ones to find customers and > to > > make sure they bring added value. > Yes of course. but they're sure that the customer will buy from them. > normally the user will buy directly from the hw maker (and this's ok) > if the hw maker allows that, since in this way the user thinks that > going directly to the manufacturer they'll have better support and > better price. I know that is can not be the real truth, but is how's > perceived from an enduser pow. > We're Digium resellers, but some .it people buy the card from other > countries (because not imposing list prices allows resellers to do > market dumping) or even direlcty from Digium. > And we apply the very same Digium list price. and the import taxes > are payed by our reseller discount. > So when the enduser buys directly from .usa, they will pay list price > plus taxes, so more than our final price. But this is not considered, > seems. > > > I don't know what's unusual in this approach? > everything. > > Matteo > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma VS. Digium
Matteo, I don't know much about DIgium, but I am comparing the distribution policy with what exists elsewhere in the market and other sectors. Digium do sell online and so many other of their resellers do. The important point is that they don't sell lower cost than their resellers, which is the case. Reseller added value is find customers and retail locally in his place with local variables of config, ...etc. They are the ones to find customers and to make sure they bring added value. I don't know what's unusual in this approach? Please elaborate. Thanks, Selon Matteo Brancaleoni <[EMAIL PROTECTED]>: > Hi, > > > Digium, the service is problematic. Well, I believe that Digium should > > services it's channels, the channels should support the resellers and the > > resellers should support > > The customers. I don't think that any company, no matter what its size or > > function is, could support the end users. Even the mighty ugly M$ has > > country based support > > Centers. > > I hate to say that, but the problem is that Digium doesn't do this. > They allow resellers to do market dumping, by not imposing fixed > list prices to resellers, they also compete with they're own > distributors/resellers by offering the cards online and by offering > services directly to end users. > In this way they're destroying they're own reseller network > and there's no commercial gain into supporting the end user > (as resellers). > > Sangoma doesn't do that. they don't sell directly, thus allowing > resellers to have a money gain and pay the time to support the end > user. > > again, I hate to say that, but is a common pow. > I hope that digium will change their mind in the way > they sells hw/services. > > Matteo > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Access Voicemail From Outside
You can use a password protected DISA functionality. Selon Bill Ford <[EMAIL PROTECTED]>: > I'd like to see what some of you are doing to reliably aess > voicemail from an outside line. > > Thanks > Bill > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how can i connect a cost display on asterisk
If it helps, then great ! This is what we do in Africa for some callshops that do not want to pay $$$ in billing software licenses. While talking, I wonder whether the field of "CAller ID" which is displayed in the IP Phone can be updated while conversation is ongoing, say every 10 seconds. In which case, may be the cost can be displayed in the content of caller ID (since at this time, people are calling and not being called)? Selon Han van Hulst <[EMAIL PROTECTED]>: > That is a fast solution. I did not know this. > But is it not possible to send messages to a SIP phone. > > But thanks for the TIP of the call meter. > > Johannes > > > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > Sent: quarta-feira, 6 de abril de 2005 10:36 > To: Asterisk Users Mailing List - Non-Commercial Discussion; Han van Hulst > Cc: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] how can i connect a cost display on asterisk > > Johannes, > > I would be curious to know if there is a solution for this. Another solution > is > that you buy a "call meter". Which is a small box that can be placed in > front > of phone phone and that can display costs. > > FXS--> call meter --> analog phone > > This call meter needs to be programmed with a table inside and a rate for > each > destination. > > > Selon Han van Hulst <[EMAIL PROTECTED]>: > > > Is it possible to connect a display that shows the costs of a call in > > progress? > > > > > > > > Can I also send the call cost to a grandstream display? > > > > > > > > Greeting > > > > > > > > Johannes > > > > > > > > > > > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how can i connect a cost display on asterisk
Johannes, I would be curious to know if there is a solution for this. Another solution is that you buy a "call meter". Which is a small box that can be placed in front of phone phone and that can display costs. FXS--> call meter --> analog phone This call meter needs to be programmed with a table inside and a rate for each destination. Selon Han van Hulst <[EMAIL PROTECTED]>: > Is it possible to connect a display that shows the costs of a call in > progress? > > > > Can I also send the call cost to a grandstream display? > > > > Greeting > > > > Johannes > > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip <-> oh323 / real-time / g729 - one way audio
Are the h323 devices on public IP or behind NAT? Selon Shaoul Jacobson - TELLINK <[EMAIL PROTECTED]>: > > > Hi, > > I am using real-time, oh-0.7.2, G729 > > Calling from (SIP)UA through asterisk towards h323 devices or the other way > round, I get only one-way audio. > > Called party can only talk, caller can only listen. > > Calling SIP to SIP is ok. > > All devices are on official IP addresses. > (no NAT) > > > Regards, > > Shaoul Jacobson > Senior VoIP Consultant > Tellink > Tel : +32 3 201 96 36 > Fax : +32 3 227 09 81 > e-mail[EMAIL PROTECTED] > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Voice mail with CCM
Let me ask a silly question, Is Cisco call manager free software ? Selon Nathan Alberti <[EMAIL PROTECTED]>: > > I did some work on my configuration to get it working and have now > documented it somewhat here: > http://voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Express+Integration > > Thanks to the list member who pointed me in the direction of the > redirect ip2ip which solved my problem. > > Regards, > > Nathan. > > > Nathan Reeves wrote: > > Anyone running Cisco Call Manager and using Asterisk for voice mail > > services? Things working well or is the concept a bit of a hassle to > > implement? > > > > TIA > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Customized ring tones
Hello, I have very few knowledge of this area. But mobile operator in belgium has launched "Hello Tone". So if you call somebody on mobile network, you can listen to U2 music, or Bethoven (chosen by callee) while the phone is ringing, even if you call from different network operator. Before I see this, I thought that phone ring tones are standards. As this can be done, can it be implemented in Asterisk if interconnected through ISDN to a carrier during incoming calls? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma VS. Digium
What about pricing of the Sangoma compared to Digium, is it comparable? Can Sangoma card handle modem data incoming calls at all? Selon mattf <[EMAIL PROTECTED]>: > Hello, > > I need to correct myself on one of the points I made in my reply last night. > As a very polite developer from Sangoma stated to me(with evidence I might > add)they have in the past and continue to today contribute code to GPL > Asterisk. It doesn't say so on their website but their developers have been > bug-checking, patching and contributing new code to Asterisk for some time > now. They just started directly giving credit from Sangoma for some of these > contributions in the bugtracker starting this week. While it is true that > they probably don't have as many full-time dedicated Asterisk developers as > Digium does, a portion of a Sangoma AFT card purchase will go towards > further development of Asterisk. So you can feel a little less-bad about > buying those Sangoma cards now. > > MATT--- > > -Original Message- > From: Brian Capouch [mailto:[EMAIL PROTECTED] > Sent: Thursday, March 31, 2005 2:43 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Sangoma VS. Digium > > > Isamar Maia wrote: > > Technically speaking not. But Sangoma's support seems to be pretty much > > better. > > > > My understanding is that to an extent when we buy Sangoma we're putting > the dagger to Digium. They're glad to use Asterisk as a selling point > for their hardware, but unwilling to donate anything back to the > Asterisk community. > > I'll be glad to stand corrected, but if that assertion is in fact true, > we should be careful to do things that actually damage Digium's ability > to leverage their development of Asterisk with their hardware sales. > > FWIW. > > b. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended GSM gateway
Hi, I don't know much about Orion brands. In Europe, most robust brands are: www.2n.cz and www.vierling.de Vierling is richest but most expensive. It does SMS callback, sms2mail, mail2sms. Regards, Selon cmould <[EMAIL PROTECTED]>: > My client is looking fro a GSM gateway (24 ports). Any recommendations > from the list. Anyone hase experience with the Orion GSM gateway? > > Also any experience integrating with a Meridan Option 51c PBX. Dont want > to reinvent the wheel. > > I also thought I might put asterisk between the Meridian and the GSM > gateway to provide authentication and call accounting, or even call > quota's. You coments appreciated. > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spandsp compilation error
Did you install libtiff libraries & prerequisites before compiling It may be an issue with your LD_LIBRARY_PATH as files do not seem to be found. Selon Dennie Verstrepen <[EMAIL PROTECTED]>: > Hello everybody, > > I'm trying to receive and sending faxes with asterisk using spandsp. But > while compiling the spandsp0.0.2pre11 (tried also spandsp0.0.1), I get > following errormessage: > > gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -c t4.c -Wp,-MD,.deps/t4.TPlo > -fPIC -DPIC -o .libs/t4.o > In file included from spandsp.h:42, > from t4.c:71: > spandsp/arctan2.h: In function `arctan2': > spandsp/arctan2.h:51: warning: type mismatch in implicit declaration for > built-in function `fabs' > t4.c: In function `t4_rx_end_page': > t4.c:566: `COMPRESSION_CCITT_T4' undeclared (first use in this function) > t4.c:566: (Each undeclared identifier is reported only once > t4.c:566: for each function it appears in.) > t4.c: In function `t4_rx_init': > t4.c:915: `COMPRESSION_CCITT_T4' undeclared (first use in this function) > t4.c:923: `COMPRESSION_CCITT_T6' undeclared (first use in this function) > t4.c: In function `t4_rx_start_page': > t4.c:972: `COMPRESSION_CCITT_T4' undeclared (first use in this function) > t4.c:974: `TIFFTAG_T4OPTIONS' undeclared (first use in this function) > t4.c:983: `COMPRESSION_CCITT_T6' undeclared (first use in this function) > make[2]: ** [t4.lo] Erro 1 > make[2]: Leaving directory `/usr/src/spandsp-0.0.2/src' > make[1]: ** [all] Erro 2 > make[1]: Leaving directory `/usr/src/spandsp-0.0.2/src' > make: ** [all-recursive] Erro 1 > > Can anyone tell me what I'm doing wrong? I'm using Debian 3.0r3 with kernel > 2.6.6 > > Thanks in advance, > > Dennie > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 gateway thru NAT
This is possible. But success depends also on whether the router can do port forwarding and whether the H323 Gateway supports NAT. This is possible with Quintum for instance with some port forwarding rules on router level. Selon VoIP Newbie <[EMAIL PROTECTED]>: > Hi all, > > I am wondering if chan_oh323 or chan_h323 supports NAT traversal the > following setup: > > H323 phone -> Asterisk ---> NAT router -> H323 gateway -> > PSTN > > I am trying to register a H323 gateway through a NAT to Asterisk for > outgoing calls to PSTN. > > How can I achieve the above? Please help and advise. > > Many Thanks. > Newbie > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallingCard Application
You can try AreskiCC at areski.net Selon chawki hammoud <[EMAIL PROTECTED]>: > I appreciate any recomendation of a simple CallingCard > Application and resources of users manual. > > > > __ > Do you Yahoo!? > Yahoo! Small Business - Try our new resources site! > http://smallbusiness.yahoo.com/resources/ > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio when h323 calls are incoming
For this, it works randomly. I have decided to work on SIP and forgot about h323 with Asterisk. I spent nights and nights trying to figure out how it works, but decided to move on. Now we are running SIP, things are better. Selon Chetan Sarva <[EMAIL PROTECTED]>: > Did you ever find a solution to this problem? > > [EMAIL PROTECTED] wrote: > > >All, > > > >I have tried very hard to make asterisk work with h323 but still strying: > > > >I have been successful making this work > > > >SIP --> Asterisk --> h323 --> termination ; > > > >But the following: > > > >h323 --> asterisk --> h323 --> Termination : works , call set up is ok but > then > >no audio is applied .There is no NAT here at all are public. > > > >I also tried > > > >h323 --> asterisk --> SIP --> terminatino: I have same problem here, audio > > > >I use g723 codec (passthrough ) > > > >Can anyone advise what is to look or is it meant not to work anyway ? > > > > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Center software opensource or commercial
Thanks Kevin for this info, If we want a box that can perform 60 calls. What would be apoproximate budget for that using AMD x86-64 ? µSelon "Kevin P. Fleming" <[EMAIL PROTECTED]>: > Erick Perez wrote: > > And what people are using to deploy super servers with astersik? > > Itanium with linux? clusters of itanium with linux? or some RISC > > processor with some *nix? cause it seems asterisk is only 100% > > supported on Linux/Intel > > or am i totally wrong? > > The highest-performing "standard" hardware to run Asterisk on today > would be quad/octal Opteron (AMD X86-64) boxes. > > In fact, hardware like that will very likely outperform the Altix system > that Signate did their benchmarking on, for quite a lot less money. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Voicemail SMS Alert - Possible?
Another option is to send sms by mail. 1-) You subscribe to an sms provider who can allow you to do mail2sms; 2-) You send sms message under the form [EMAIL PROTECTED] ; 3-) SMS provider receives SMS from you and will send it through its gateway; Hope this helps Quoting Marco Ziglioli <[EMAIL PROTECTED]>: Use externnotify (see http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf) with a script to send sms. Some time ago I used a perl script called sendSms found in Internet. Bye. Marco -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Julius Kidubuka Inviato: lunedì 14 marzo 2005 09.09 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Voicemail SMS Alert - Possible? I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? And if it is, what do I need to do to get the service up and running? Ideas are most welcome. Thanks, Julius. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Avoiding connect signal in two stage dialing
Hello,
I am using asterisk for two stage dialing:
1-) I make a call from my voip phone;
2-) Asterisk dials first a access number, iputs a PIN and then dials destination
number;
However, I am getting the connect signal from the moment access number connects.
I would like to avoid receiving this signal and only get connect signal after
the destination number connects.
Any clue how to implement this?
I have been using this:
exten --> _. , 1, Dial(SIP/myprovider.com, 30,
D(www447881234ww12345678www${EXTEN})
This message was sent using IMP, the Internet Messaging Program.
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RE: [Asterisk-Users] Print-to-Fax client
This conversation is interesting, What about a driver that will send the print out to Asterisk, on the same network to be sent as Fax ? Is there anything that already exists for this? Quoting Florian Overkamp <[EMAIL PROTECTED]>: Hi, -Original Message- You should be able to download one (for WIndows and possibly Mac) from efax or j2.com I think. http://www.efax.com/en/efax/twa/page/download?rqcp=2 http://www.j2.com/jconnect/twa/page/download You might be able to do that, but take a good look at the license agreement on the driver - you might not be allowed to use the software fully without having a subscription to their services. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Software SIP fax client
Check: http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html What you need is convert the document you need to send into a tiff file and put it in a directory. Which I believe is functionality you want to achieve. Selon Justin Newman <[EMAIL PROTECTED]>: We have something on the way. Regards, Justin Newman [EMAIL PROTECTED] - Original Message - > Does anyone know of a software SIP fax client? Something I can install = > on a PC which connects to the asterisk server and sends/receives faxes? = > Something like XLite - but to fax instead of to phone. > =A0 > I know of the "fax machine connected to an ATA" solution, but that's not = > really what I'm looking for :-) > =A0 > Thanks > -Manuel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No audio when h323 calls are incoming
All, I have tried very hard to make asterisk work with h323 but still strying: I have been successful making this work SIP --> Asterisk --> h323 --> termination ; But the following: h323 --> asterisk --> h323 --> Termination : works , call set up is ok but then no audio is applied .There is no NAT here at all are public. I also tried h323 --> asterisk --> SIP --> terminatino: I have same problem here, audio I use g723 codec (passthrough ) Can anyone advise what is to look or is it meant not to work anyway ? This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Process incoming faxes in Asterisk
Hello All I am looking for a solution that can do this: 1-) Receive incoming fax; 2-) Read content and identify a zone in the fax where there is a hand written name; 3-) Based on name, query a database; 4-) Act based on the result in the database; I understand asterisk can receive fax and redirect it in PDF format. Are there any asterisk users who know if such solution already exist or help where to get it working ? Any help on this is much appreciated ! Best regards, Hakem, This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incoming h323 calls, routed to SIP/H323 drop after connection
Hello, I am attempting to use Asterisk as a protocol converter. I have set up asterisk to route incoming h323 calls to a SIP termination carrier. I make a test, call is coming correctly, is rerouted to termination carrier. Call connects and phone rings. Then, I pick up the phone and it hangs up after 2 seconds. I initially thought it was a codec issue. I made sure codec is g729 in all sip.conf & h323.conf parts (general context + specific contexts). Still, call drops after connects and gives error "cannot bridge between X call and Y call". Is this familiar to anyone? Do you have idea what to search next? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk performance monitoring
I'm not sure it answers all your questions but there is ast-stats from http://areski.net/areski/index.php? option=com_content&task=category§ionid=5&id=70&Itemid=54 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP port blocked in Dubai ?
Hello, We had same problem in other african country. We could resolve it through using IAX Bridge in Asterisk since it only uses one port of yoru choice. For your solution, you need: 1-) Scan outgoing / incoming open ports by your ISP; 2-) If there remains many open ports, you may still run SIP by changing ports; 3-) Alternately, you need to get IAX bridge working if need be; It is my understanding the Government considers VoIP illegal since it would compete with the Government run service in Dubai. This is based on a conversation with a coworker who was raised there and just returned from a vacation home. You should check this out to make sure my facts are correct. Manjit Riat wrote: > Does any know if SIP ports are blocked in dubai (UAE)? Anyone in UAE > using FWD or similar services and connecting to SIP proxies in US? > > > > Thanks. This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] incoming calls in h323 do not come to right dialplan
Soren, I tried the variable UserByAlias=no and it worked for me. Thank you very much for this note ! Selon Soren Rathje <[EMAIL PROTECTED]>: [EMAIL PROTECTED] wrote: Good to hear I am not alone. Actually, I am using the Nufone's h323 module. Still this creates the problem. I had a braod look at the code and it seems that it is not possible that incoming calls go to other places than "general" context (I am not sure I understood it all, but almost). I had a similar problem, my Siemens OptiPoint 300 A refused to send calls to the corresponding context, so after a short review of the code I found the parameter "userbyalias" and included "userbyalias = no" in h323.conf and now chan_h323 use the IP address to match, not the H323_ID. /Soren ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] incoming calls in h323 do not come to right dialplan
Good to hear I am not alone. Actually, I am using the Nufone's h323 module. Still this creates the problem. I had a braod look at the code and it seems that it is not possible that incoming calls go to other places than "general" context (I am not sure I understood it all, but almost). So, one workaround I found is this: 1-) Authenticate incoming calls through caller id. So in the "default" or "general" dialplan, you will move calls coming from specific callerID to the context of your choice. 2-) All other calls are automatically handled with default option. So, this resolves the security issue at least. I will keep you updated if I can manage to find a solution from a code point of view. Any experience to describe here is highly appreciated. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Caleb Sent: Monday, February 07, 2005 5:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] incoming calls in h323 do not come to right dialplan I am also facing the same problem, running on chan_h323 and CVS-HEAD-12/17/04-15:07:40. Anybody managed to solve this? Cheers On Mon, 7 Feb 2005 07:10:55 -0800, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Hello, > I am moving topic from asterisk-dev list to asterisk-users list. Did > anyone succeed receive incoming calls in h323 and orient them to right > context based on "host" identification? > > To summarise, I have quintum Gateway sending call to Asterisk box, and > I would like to use asterisk as a protocol converter h323 --> sip. > > in h323.conf, I have > > [quintum_gw1] > type=user > host=192.168.1.210 > context=outbound > incominglimit=2 > disallow=all > allow=g723.1 > > However, when asterisk receives call from this box, it does not send > it to context "outbound" b ut the "general" context and call fails > because it does not have instructions in this context. > > Did anyone successfully get incoming h323 call? > > What is my error ? > > Thanks in advance, spent too much time on this ! > > > This message was sent using IMP, the Internet Messaging Program. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incoming calls in h323 do not come to right dialplan
Hello, I am moving topic from asterisk-dev list to asterisk-users list. Did anyone succeed receive incoming calls in h323 and orient them to right context based on "host" identification? To summarise, I have quintum Gateway sending call to Asterisk box, and I would like to use asterisk as a protocol converter h323 --> sip. in h323.conf, I have [quintum_gw1] type=user host=192.168.1.210 context=outbound incominglimit=2 disallow=all allow=g723.1 However, when asterisk receives call from this box, it does not send it to context "outbound" b ut the "general" context and call fails because it does not have instructions in this context. Did anyone successfully get incoming h323 call? What is my error ? Thanks in advance, spent too much time on this ! This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g723.1 voicemail/conference files segfault *
Thanks Matteo,
Now I have a backtrace if that will help. I am not a programmer and this
really means nothing to me. I can only tell you that I have a g723.1 encoded
file (conf-onlyperson.g723) in /var/lib/asterisk/sounds/ when this happens.
#0 0x08058291 in ast_write (chan=0x8111718, fr=0x) at
channel.c:1332
1332switch(fr->frametype) {
(gdb) backtrace
#0 0x08058291 in ast_write (chan=0x8111718, fr=0x) at
channel.c:1332
#1 0x0805de8b in ast_readaudio_callback (data=0x8111718) at file.c:508
#2 0x0805f09d in ast_streamfile (chan=0x0, filename=0x41e42608
"conf-onlyperson",
preflang=0x8111718 "SIP/7600-894e") at file.c:575
#3 0x41e41733 in conf_run (chan=0x8111718, conf=0x80cd398, confflags=0) at
app_meetme.c:246
#4 0x41e40a73 in conf_exec (chan=0x8111718, data=0x8111718) at
app_meetme.c:585
#5 0x08060eca in pbx_exec (c=0x8111718, app=0x80f74a8, data=0x49dbcd2c,
newstack=1) at pbx.c:388
#6 0x08067ef8 in pbx_extension_helper (c=0x8111718, context=0x41e42608
"conf-onlyperson",
exten=0x8111908 "", priority=1, callerid=0x80e42e0 "7600",
action=1105480556) at pbx.c:1130
#7 0x08062d2c in ast_pbx_run (c=0x41e44b6c) at pbx.c:1614
#8 0x080685b1 in pbx_thread (data=0x) at pbx.c:1830
#9 0x400252b6 in start_thread () from /lib/tls/libpthread.so.0
(gdb)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brancaleoni
Matteo
Sent: Tuesday, July 15, 2003 8:29 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] g723.1 voicemail/conference files segfault *
>
> Off the topic: where can I find the core dump? I am running asterisk on
> Redhat9.
in the dir where you started *. but you must have to issue
'ulimit -c unlimited' if you wanna asterisks dump cores.
if you're starting it via the init.astersik script, you will found
the cores in /tmp/
Matteo
--
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web : http://www.espia.it
Phone : +39.02.70633354 - ext 911
IAX(2): [EMAIL PROTECTED] - ext 911
or tel:17005662458 - ext 911
___
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[Asterisk-Users] g723.1 voicemail/conference files segfault *
Hi, First of all I am not sure that what I am trying to do is correct/supported, but here is what I'm trying to test: Some of my endpoints only have g723 codecs. Because of this I am only allowing g723.1 codec in sip.conf and h323.conf. Calls between endpoints work fine. I am trying to configure voicemail and meetme applications. I see that all voice files in asterisk are in gsm format and when I try to place a call to the voicemail/meetme I get some message saying: File channel.c, Line 1399 (ast_set_write_format): Unable to find a path from 2 to 1 This probably means that GSM<->G.723.1 transcoding is not supported (which is normal). However, when I try to use a voice file pre-encoded with G.723.1 codec (for example conf-onlyperson.g723) I not getting the " Unable to find a path from 2 to 1" message, but Asterisk segfaults. My guess is that asterisk is probably capable of playing/streaming files to g.723.1 endpoints if the files to be played are already encoded with g723 codec. Right? Is this feature supported first of all and has someone already tested voicemail/meetme apps with different voice files (.g723 for ex.)? Off the topic: where can I find the core dump? I am running asterisk on Redhat9. H. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Use dialing plan from h.323 gatekeeper?
Hi, I think I understood how to achieve this. Anyway, a working config is welcome if anyone has already done it. hristo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of HT Sent: Wednesday, July 09, 2003 2:54 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Use dialing plan from h.323 gatekeeper? Hi, I want to configure * to use a gatekeeper for routing calls to H.323 endpoints. I imagine it will work like that: * (chan_h323) will query the gatekeeper where to terminate the dialed number and the gatekeeper will return the information for the h.323 gateway. after that chan_h323 will try to make the call to the gateway it has received from the gatekeeper. so instead of duplicating a gatekeeper's dialing plan (with all registered gateways and terminals) in *, can I just configure * to receive the endpoint for terminating a call from the gatekeeper for lets say all calls starting with prefix 9? Hristo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Use dialing plan from h.323 gatekeeper?
Hi, I want to configure * to use a gatekeeper for routing calls to H.323 endpoints. I imagine it will work like that: * (chan_h323) will query the gatekeeper where to terminate the dialed number and the gatekeeper will return the information for the h.323 gateway. after that chan_h323 will try to make the call to the gateway it has received from the gatekeeper. so instead of duplicating a gatekeeper's dialing plan (with all registered gateways and terminals) in *, can I just configure * to receive the endpoint for terminating a call from the gatekeeper for lets say all calls starting with prefix 9? Hristo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to make * send RTCP reports
Thank you for the answer. Anyone working on that? I am trying in the meantime to disable the RTCP reports on the gateways, hoping that it will work like that. hristo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer Sent: Saturday, July 05, 2003 9:52 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] How to make * send RTCP reports > I can see on the console that * is detecting incoming RTCP reports so there > should be some RTCP functionality in it (although I have seen a message from > February saying the opposite). My question is if/how can I make * send RTCP > report to the vocaltec gateways. I think any RTCP packet will do the trick > as long as the vocaltec gateway gets it on a regular basis (I don't care if > the information in it is correct). It would require some additional code, but could be done. Some of the basics are in place now but not nearly enough code, as it did not end up being required for video as we had though tit would be. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to make * send RTCP reports
Hi, I am plying with * for 10 days now. I am testing with a couple of vocaltec h.323 gateways (FXO and PRI) cisco ata-186 (configured for SIP) and MSN messenger (SIP). They all seem to interoperate. However I have a problem when * is sending calls to the vocaltec gateways. Vocaltec gateways are monitoring the RTCP reports send from the remote gateway (in this case *) and if they don’t get a report for 60 seconds they will disconnect the call (assuming internet disconnection). Because of this all my calls have duration of one minute. I can see on the console that * is detecting incoming RTCP reports so there should be some RTCP functionality in it (although I have seen a message from February saying the opposite). My question is if/how can I make * send RTCP report to the vocaltec gateways. I think any RTCP packet will do the trick as long as the vocaltec gateway gets it on a regular basis (I don’t care if the information in it is correct). H.
