[Asterisk-Users] Specifying different SIP packet destination from hostname in request line?
I'm stumped. I'm trying to configure my Asterisk setup to use a new embryonic UK ITSP (who wants to remain nameless for the moment ...). They supply their own UA, which works fine. But Asterisk is proving to be a problem. Ethereal traces show that their UA is sending SIP packets direct to an IP address aaa.bbb.ccc.ddd, but that the address in the REGISTER and INVITE methods is specifying a hostname, eg: REGISTER sip:[EMAIL PROTECTED] SIP/2.0 and INVITE sip:[EMAIL PROTECTED] SIP/2.0 Unless I've missed something, there doesn't seem to be a way to tell Asterisk to use a different destination for the SIP packet to that specified in the method. Is there a way to accomplish this? Thanks -- David Gurr Congruity Ltd. Hemel Hempstead UK ___ 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] Legacy Toshiba Phones
Leo wrote: Not necessarily so. Recently I discovered that Artisoft's Televantage Soft PBX can support Toshiba Strata CS digital phones (DKT 2000 and 3000) through a PCI 16-port digital station card (Toshiba part #CS-DKTU-TV). Apparently, the Strata CS is an OEM licensed version of Televantage. It would be quite cool if an Asterisk driver can be developed for the 16-port digital station card. Interesting. I just checked out the Televantage site at TrueData (http://www.truedataonline.com). In their FAQ there's a question Can TV (Televantage) use Digital Sets?. The answer includes the tidbit: The Toshiba digital station cards are a slight variation of the Intel MSI160PCI and are interoperable with other Dialogic - Intel Televantage hardware. The good news is that Intel have Linux drivers for the MSI160. Guess someone needs to find some details on how the Tosh card differs ... The bad news is that the Tosh station card doesn't come cheap ... a quick google search shows prices around $2,500! For 16 ports? Ouch! At that price, it's cheaper to throw the Tosh phones away and buy IP hardphones. -- David Gurr Congruity Ltd. Hemel Hempstead UK ___ 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] Any UK PipeCall/PipeMedia users?
Has anyone out there used the PipeMedia/PipeCall PSTN gateway? Anything good/bad to say about it? I'm considering using them for a new customer. They seem to have good rates, good provisioning tools and (better still) give commission on usage to dealers. -- David Gurr Congruity Ltd. Fax: 0871 661 1756 Hemel Hempstead UK ___ 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] ISDN BRI card exepriences in UK
Looking for folks experiences with ISDN BRI cards in the UK ... what's good and what's bad and any gotchas. Thx -- David Gurr Congruity Ltd. Hemel Hempstead UK ___ 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] FXO interfaces used in UK?
What FXO interface methods are folks using successfully in the UK? I'm looking for good, known-to-work solutions for commercial use for two PSTN trunks on an Asterisk box. Here's the options I have, as I see it: i) Two Digium X100Ps. Pro - cheap (c. £120), CE approved. Con - UK line impedance mismatch, with resulting echo problems, plus needs two PCI slots. ii) Digium TDM400P with two FXO modules. Pro - still fairly cheap (c. £200), and can be set to UK line impedance, plus only uses one PCI slot. Con - Not yet CE approved. iii)Voicetronix Openline 4. Pro - reasonable price (c. £310), CE approved, only uses 1 PCI slot. Con - UK line impedance mismatch. iv) FXO gateway: - Multitech MVP210. Pro - UK line impedance, no PCI slot needed, good local technical support, CE approved. Con - expensive (£590 list) - Mediatrix 1204. Pro - As above, plus not so expensive (£389 from Telappliant). Con - can't get to see manuals to check functionality until you buy it I'm leaning towards either the TDM400P with 2xFXO, or the Mediatrix 1204, though because of the cons I'd like to hear from folks that have successfully used these in this type of configuration in the UK before I shell out for them! -- David Gurr Congruity Ltd. Hemel Hempstead UK ___ 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] Multiple SIP phones ringing for same extension
Can someone confirm what I should expect the correct behaviour to be on incoming calls if I have multiple SIP phones configured for the same username? I'd expect all the phones registered under the username that that extension is associated with to ring, and the first one that answers gets it. What I get, is just the first phone that registered gets a ring. The second one doesn't ring at all. -- David Gurr Congruity Ltd. Hemel Hempstead UK ___ 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] Sound file quality
I'm building a phone-in demo system to use for introducing Asterisk to prospective clients. One of the things I'm wary of is their likely preconceptions that VoIP systems will have poor audio quality. As a result, I'd like to ensure that the voice prompts I'm using have the best possible audio quality. Is it possible to use sound files at higher than 8kHz sampling? My callers will be coming in over PSTN to a VoIP gateway and then to me by uLaw/aLaw ... would higher sampling rates gain me anything in this configuration? -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ 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] How do folks handle NAT routing?
I'm interested to hear how folks are handling NAT SIP routing issues in the wild for commercial use. Are you using a commerical SIP-aware NAT router solution? If so, what? Are you using a software SIP-proxy like SER or siproxd? If so, which? Do you set everything to canreinvite=no in sip.conf? Any comments about real-world implementations would be welcome. Thanks -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ 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] RE: No incoming audio on incoming SIP calls
Solved my own problem ... thought I'd record it here for any others who come across it. The problem arises since Asterisk is trying to get out of the way of the media stream, by doing a SIP re-INVITE to get the two ends of the conversation to talk directly. This won't work, as Asterisk is telling the calling party that the IP address to talk to is the private IP address of the softphone on the internal network. Adding canreinvite=no to the softphone's stanza in sip.conf solves the problem. It would be helpful if Asterisk noticed that it's about to tell the other end to use a private IP address ... the ranges are well known, and Asterisk could do an implicit canreinvite=no in this situation. The same problem didn't occur on outgoing calls as the Dial string includes a t for timeout - as per the wiki, this means that Asterisk must stay in the stream to be able to implement this. Of course, the other way to solve this would be to use a proper SIP proxy server which handles RTP stream port forwarding ... something I must get around to. -- David Gurr Congruity Ltd. Hemel Hempstead, UK -Original Message- From: David Gurr [mailto:[EMAIL PROTECTED] Sent: 04 August 2004 14:05 To: [EMAIL PROTECTED] Subject: No incoming audio on incoming SIP calls Now this is really frustrating. Everything was working fine, and now it isn't ... I don't think I've changed anything that would affect this, but I guess you never can be too sure. My setup is as follows: SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall box. This is all on the internal network. Asterisk then dialing out through various means - SIP to Stanaphone, FWD, Gossiptel and PSTN via an X100P. For incoming calls, an 0870 number from CallUK routes to my FWD account, and an 0870 number from Gossiptel routing to my Gossiptel account. Outbound calls all work fine ... I get audio in both directions, no problem. Incoming calls on either 0870 number connect fine, and audio goes from the softphone to the caller, but not the other way ... I hear no audio on the softphone from the caller's phone. I'm getting no alerts from my firewall that it's dropping anything. I know my way around packet sniffers, but I don't know what to look for here. What should the inbound audio packets look like? Thanks -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ 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] Can Zap detect line is already off-hook?
I have the need for a slightly odd * configuration for testing purposes. I have a working * setup with SIP softphones, VoIP trunks and a single X100P clone for PSTN access. The PSTN line I'm using for testing is also in use by other folks. For incoming calls, I'd like to set is up so that * functions as a voicemail backstop on this line. This much is working fine. For outgoing, I'd like * to be able to tell if the line is already in use by someone else - ie when it tries to take the line off hook, can it detect that the line is already off hook and return to the dialplan that the line is unavailable? -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ 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] UK VoIP-PSTN gateway recommendations
I'm looking for recommendations for UK-based VoIP-PSTN gateways. They should ideally offer: - IAX connection - Multiple simultaneous calls on a single account - Lower call rates than BT Business - Auto-top up or invoicing in arrears I can find several that offer one of these facilities, but none that offer all. Thanks! -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ 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] Selling asterisk-based solutions
I'm curious as to folks experiences in selling asterisk-based solutions. In sales-speak, what are the common compelling reasons to buy? I can think of the following potential ones, but I'm keen to find out what seems to work in practise: - Customer wants to cut cost of calls, implements * and signs up to a VoIP/PSTN gateway - Customer wants a new PBX but doesn't want to pay typical PBX prices - Customer has a distributed user base across offices and homes, and wants to cost effectively integrate them into one PBX - Customer wants flexibility in configuration - needs might change rapidly Thanks -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ 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] Stripping characters from SIP dial strings
I'm having problems in dialing numbers over SIP that include characters from the UK international phone number conventions. I have my contacts in Outlook, with the numbers represented as: +countrycode (area code) numberpart numberpart eg: +44 (20) 7834 1234 or: +1 (801) 555 1234 I'm using the SJphone softphone, doing my testing through the Stanaphone SIP/PSTN gateway, and have Asterisk in the middle working fine for everything else. SJphone can pick up the numbers from Outlook, and send them to Asterisk, but Asterisk is finding that Stanaphone is barfing on any number that includes a space or a ( or ). If I manually enter the number without spaces or parentheses, it's fine. The following questions arise: - Are these characters allowed in a SIP URI? - If so, is this an Asterisk problem, a Stanaphone problem, or an SJphone problem? - How can I simply strip out spaces or parentheses from a dialed extension string? Thanks -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ 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] How to allow softphone to dial thru with full SIP URI?
I'm using the SJphone softphone, and I've got a nice little SIP-only setup, using (amongst others) stanaphone, VOIPtalk and FWD. But I'd like to be able to use my SJphones to dial directly to folks who provide a SIP URI, eg: [EMAIL PROTECTED], without either having to switch profiles in SJphone (to direct SIP) or having to define calluk.com (in this example) as a destination in extensions.conf. Is this possible? The less user twiddling to get a call through, the better ... -- David Gurr Congruity Ltd. ___ 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
