Re: [Asterisk-Users] Some questions (maybe Nikotel related)
inline On Jan 10, 2005, at 10:12 PM, Christian Peter wrote: - If I call outside (with Nikotel to German Telekom) there is a remote hangup after 2 minutes. I've seen other people posting this but nothing helped. I luckily managed to get around this issue with the following workaround: The provider section should only contain disallow=all and then only allow=gsm. If I add allow=alaw . After 60 seconds nikotel send a reINVITE to your box. If your box does not respond then the call gets cleared after 120 seconds. I do not know why this is up to the codec order of * - I sniffed the traffic and came to another strange issue. From time to time asterisk sends a OPTIONS packet (even before REGISTER). This Seems that * keeps routers WAN port this way packets have a From header which looks like this: sip:[EMAIL PROTECTED] Nikotel does of course not recognize this address and sends a Call leg or transaction does not exist. Is this a bug or intended behaviour? Looks like the OPTIONS request happen outside of an dialog. - No internal Nikotel call (phone number beginning with 99) reaches my friends (which have similar sip.conf and extensions.conf). Somewhere I read that the section must be named like the host calamar0.nikotel.com so that asterisk finds it. It didn't help. Did someone manage to get this working? There is(should be) a 302 Response fix in the current CVS Michael ___ 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] BugetTone Bug Showstopper,
Use the 'send' button On Jul 29, 2004, at 3:26 PM, Kanuri, Seshu wrote: I have setup Grandstream to connect to my Asterisk Server. All the digits 0-9 are accepting dtmf. But When I try to send the call by Pressing # Key, nothing happens. Does anyone has a standard configuration for Asterisk and Grandstream as a PDF file or something to see. How do you send the connect signal? Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Foster Sent: Thursday, July 29, 2004 9:13 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] BugetTone Bug Showstopper, On Wed, 28 Jul 2004 23:31:06 -0400, Seth Remington [EMAIL PROTECTED] wrote: On Wed, 2004-07-28 at 21:00, James Gardiner wrote: How do I get Asterisk to recognise the # key from the granstream phone for doing transfers? Make sure the Grandstream is configured to send DTMF via SIP INFO instead of in-audio. -Seth Also, don't forget to disable the #-key as redial feature. ___ 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] SIP and RTP / 302 after 18x / Call forwarding after announce
Experts asked now: Is there a way to make this call scenario possible: After an INVITE was received at the asterisk an announcement should be played, then, the caller should be forwarded to another loc. REFER should not be used in any way! I thought about something like this: Client Asterisk --- INVITE > 183 Session Progress RTP Stream [ .. some time .. ] 302 Moved .. Contact: [EMAIL PROTECTED] ACK > But i could not figure out how to make a answer/playback happen without the final (200 ok) response to the INVITE dialog. I thought about patching the chan_sip, but this would take me away from the branch!? Please only answer if: - you know a solution (none sip REFER!) - you may have just an idea (working or not - not important :) ) Sincerely , Michael
Re: [Asterisk-Users] FINALLY! a good book about Asterisk.
It is a good resource for neck tie non-geeks in small offices and will hopefully evangelize many of the uhh, it's open source and it is for free = so this could not be good heathens. Michael On Jul 8, 2004, at 11:19 PM, usedcanon wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: 08 July 2004 20:15 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk. what does that have to do with an overpriced book? and i agree with Joe. With this book sourcing most of the documentation directly from wiki, why pay for something thats free? Id rather donate $49 to keeping wiki free to the enviroment. ___ 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 Jul 8, 2004, at 9:51 AM, Steve Kennedy wrote: On Wed, Jul 07, 2004 at 07:19:44PM -0800, rich allen wrote: what do you mean not quite right??? i[..]blocked clid CLID is NEVER blocked at the SS7 level (well almost), it flagged as withheld. Bingo, if you have a SS7 switch at the net then you can send whatever you want. Michael ___ 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] looking for newbie resources
www.voipbox.de, EN lang blog for news etc On Jul 4, 2004, at 8:53 PM, Steven M. Sawczyn wrote: Hi, I am very interested in VOIP and telephony in general, although admittedly, I don't know much about the theories and protocols behind it. ___ 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] WI FI IP phones??
i prefer zyxel p200w for a picture see http://www.voipbox.de/images/private/protzundco/equip.jpg at the upper left corner James Moran wrote: Are there any other wireless IP phones out there other then the Cisco 7920?? ___ 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: asterisk-grandstream call
Asterisk is ignoring the codec offer of the caller. Asterisk is always sending the whole codec list inside 200 OK (on invites), which should be just a subset of that what is received before within the dialog initiating invite. Workaround: Try "disallow=gsm" regards, Michael Bill Michaelson wrote: I am trying to muddle my way tthrough getting something - actually anything to work - with Asterisk. I've acquired a Grandstream phone and I've got * on a Red Hat 9 box. I've gotten to a point where I can see (via ethereal) that the phone REGISTER's successfully with asterisk, and then I try to dial into voicemail. This is what I observe in the packet trace... GS: INVITE - * *: Status 100 (Trying) - GS *: Status 200 (OK with session description) - GS Does the GS then send an ACK? It should. If it doesn't then this probably means that it hasn't received the 200 response. (firewall issue?) If it is sending the ACK, then it is probably a codec issue, as has been already mentioned. GS doesn't always seem to do very well in codec selection. Doug - Thanks for that hint. I see what you mean. When configured for FWD, the GS does indeed send an ACK at an equivalent point in the protocol. But no, the GS does not send an ACK when configured for my * box. I suppose the * box is expecting it, because about one second later, the * box resends the 200 message - this in spite of the fact that has started spewing RTP furiously. Both devices are on the same LAN, with no intervening firewall, and the OK ought to be visible to the GS (the packet even contains the expected destination MAC ID, derived earlier via ARP). That makes two mysteries: 1) why doesn't the GS seem to see the OK? and 2) why does * send the RTP stream in spite of the fact that it has not received the ACK from the GS? Shouldn't it wait? Regarding codec selection, I see a minor difference between the FWD and the local * box test cases, but I know nothing about the negotiation protocol... With FWD, the OK message lists 3 Media Formats: Media Description, name and address (m): audio 10496 RTP/AVP 0 8 101 Media Type: audio Media Port: 10496 Media Proto: RTP/AVP Media Format: 0 Media Format: 8 Media Format: 101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 But with the local box, it lists one other - note the addition of GSM... Media Description, name and address (m): audio 16708 RTP/AVP 3 0 8 101 Media Type: audio Media Port: 16708 Media Proto: RTP/AVP Media Format: 3 Media Format: 0 Media Format: 8 Media Format: 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Don't see much else different in the packets. It might also be relevant that the FWD connection, which works successfully, is through a firewall with NAT. Still fishing... thanks for your attention - much appreciate not being alone here!
Re: [Asterisk-Users] voip phones
no, it is an original. But, i prefer the black chassis one Isamar Maia wrote: not to mention, fortune cookies are included! :) Hey Chinaman... I was wondering if the following SIP phone is just a Grandstream's OEM or just a japanese copy... http://sipphone.livedoor.com/ What do you think? Isamar ___ 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 phones
I prefer SIP Phones, Grandstream BT-100 IP-Phone or the HT-286 Analogue telephone adapter Why? - brilliant user interface, with or with out a web browser - cristal clear voice even with low band codecs - PPP over ethernet (PPPoE) aware - continual firmware improvement - plenty of tweak options - economically priced - protocol conform - made in china - fast shipping Retail from $39 to $245 .. google is your friend. Tim Sailer wrote: What is the best inexpensive voip phone out there? I want to try a few with *, but don't want to go broke while I'm just playing around... Tim ___ 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-Invite between SIP phones
basically for (re)negotiation on session parameters like: media codecs, media IP and PORT In common it is useful to put a line on hold by setting the media IP to 0.0.0.0 or for a soft redirection of the media stream to another IP and/or PORT. On the second hand there is a feature called "timer" to check for aliveness of a session. Al wrote: What I would like to understand is in what situations reINVITEs are issued. Anyway, I got the following messages when trying to apply your patch. patching file chan_sip.c Hunk #1 succeeded at 160 with fuzz 1. Hunk #2 succeeded at 365 with fuzz 1. Hunk #4 FAILED at 2253. Hunk #5 FAILED at 2291. Hunk #6 FAILED at 5019. Hunk #7 FAILED at 5168. Hunk #8 succeeded at 5911 with fuzz 1. Hunk #9 FAILED at 6245. Hunk #10 FAILED at 6397. patch unexpectedly ends in middle of line Hunk #11 FAILED at 6683. 7 out of 11 hunks FAILED -- saving rejects to file chan_sip.c.rej Al --- Kannaiyan Natesan [EMAIL PROTECTED] wrote: Hi, I think canreinvite=yes won't work in most of the situations. I have implemented Redirect SIP 300 Message to redirect to the SIP address you speficy in the sip.conf. Where you can have , register = username:[EMAIL PROTECTED]/extension [extension] redirect=yes redirecturi=sip:[EMAIL PROTECTED] redirecturi=sip:[EMAIL PROTECTED] ... will make to redirect to all the URI's yu specify in the sip.conf. I'm also working on this so that it can get the redirections from the database rather than reloading asterisk all the time when you modify the redirection uri. You can check through that. http://bugs.digium.com/bug_view_page.php?bug_id=879 Message transmission is alright, but for some reason it is not working. Can you test with yours and let me know where is the problem, I will modify the code once you get the clue where is the problem on it. If successfully please send me the sip debug message and I will just make sure it works for all. Kannaiyan - Original Message - From: "Low, Adam" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:00 PM Subject: RE: [Asterisk-Users] Re-Invite between SIP phones I'd suggest placing a packet sniffer (tcpdump, etherreal) and see whats happening because it works great for me and always has but I guess it also requires support on the end-points and possibly (assuming non-cisco enviro) there maybe an option that needs to be configured on your phones/gateways. Please provide more information on your setup ... -Original Message- From: Al [mailto:[EMAIL PROTECTED]] Sent: Tuesday, January 20, 2004 2:52 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re-Invite between SIP phones Already did that, but it's not working. Al --- "Low, Adam" [EMAIL PROTECTED] wrote: canreinvite=yes within sip.conf entities ... -Original Message- From: Al [mailto:[EMAIL PROTECTED]] Sent: Tuesday, January 20, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re-Invite between SIP phones Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __ Do you Yahoo!? Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ 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 __ Do you Yahoo!? Yahoo! Hotjobs: Enter the
Re: [Asterisk-Users] SIP Asterisk - Nikotel disconnects after 1 Minute
Please add canreinvite=yes and, when the * is behind a NAT router,
nat=yes to section [nikotel]. As a nikotel customer, you can also
open a ticket and request help from nikotel.
Michael
Daniel Chabrol wrote:
Hello list!
I'm using Asterisk CVS-11/22/03-04:28:51 and try to route my normal
(classic) phone calls via nikotel (www.nikotel.com). I can talk about
1 minute and get then disconnected. Here my current configuration
parts which affect nikotel:
register = chabrol:[EMAIL PROTECTED]/500
[nikotel]
type=friend
secret=PASSWORD_REMOVED
username=chabrol
fromuser=chabrol
host=calamar0.nikotel.com
qualify=1000
context=internal
I also tried the register without /500 because there are no calls
routed inwards via nikotel and configured the type type=peer.
Additionally I tried to set auth=md5 and left off the qualify
parameter. But it changed nothing.
In the extension file i use:
[chabrol]
include = internal
exten = _00N.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Any ideas?
Best regards,
Daniel
___
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
Re: [Asterisk-Users] Streaming MOH
1. can someone please quote the text from this restricted page which is linked below to the list. could be helpful for some. 2. just for the stats, i prefer html John Todd wrote: Hi All, I keep asking things as they come into my head. Is there any way to grab an audio stream and pipe it out as the MOH? I am a helper at a local Charity Hospital Radio Station and thought it would be nice to pipe the studio output to waiting callers. Dave Dave - 1) Please don't post HTML to the list. Some people appreciate the formats less than you might think. 2) http://bugs.digium.com/bug_view_page.php?bug_id=413 JT ___ 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
Re: [Asterisk-Users] Huge silence breaks between Cisco 7960 phone X-Lite
Could be comfort noice ? Check for PT 13 or 19 Michael Ray Burkholder wrote: Huge silence breaks between Cisco 7960 phone X-Lite Does any one else have problems with huge, random silence breaks between an X-Lite and Cisco 7960 SIP phone? Both are running g.711. Softphone to/from softphone works, softphone to/from iax2 works, iax2 to/.from cisco phone works. However, voice as heard on X-Lite is just fine from Cisco, but voice as heard on Cisco from X-Lite has random silent breaks of one or two or three second duration on a very regular basis. Any ideas on how to get rid of the random silent breaks? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses dangerous content at One Unified and is believed to be clean.
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Michael T Farnworth wrote: On Fri, 24 Oct 2003, Michael Koehler wrote: :) imo it is called Budge Tone .. "budge" from "move" What will you guys think what the name "HandyTone"imply, which could suggest convenience ? Anything that gets rid of 'Budget' in the name is a good idea in my mind, as long as it isn't changed to CheapTone :-) 'Handy' is quite a good choice I think. If you tried to turn it into SuperTone it might be just viewed as being over the top and unbelievable, so you probably do need something in the middle. You could be right. "DeskTone" would be the right choise then maybe.. Michael
Re: [Asterisk-Users] Survey: Grandstream improvements.........
:) imo it is called Budge Tone .. "budge" from "move" What will you guys think what the name "HandyTone"imply, which could suggest convenience ? The BT looks better as the majority of all hotel room phone i've ever seen in the US. Dave Weis wrote: On Wed, 22 Oct 2003, Michael T Farnworth wrote: It just struck me that the easiest improvement would be to drop the name BudgeTone, because 'Budget' tends to imply cheap, which in turn suggests poor quality. The phones need a name which implies 'High Quality'. Second that. It does look/sound cheap.
Re: [Asterisk-Users] Survey: Grandstream improvements.........
10 - Alphanumeric Display. There is nothing more important. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] primuxisdn capi
it works well with chan_capi. Marian Danisek wrote: Hi, does anybody know if primus isdn cards - they support capi under linux, provided by own driver are usable with asterisk together with capi channel driver ? http://www.primuxisdn.de/primux/index.htm regards Marian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
Sorry, but my * is behind NAT and i have no problems with SIP, and it even works with NAT to NAT and without forwarding ports or similar effords. Michael Stephen Varga wrote: On Wed, 2003-09-24 at 21:50, Uriel Carrasquilla wrote: Adam: in reference to my first message, the NAT on the SIP/GS (a D-Link router) has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being forwarded to the Sip/GS. The Asterisk server, also behind another NAT (Linksys), has the same ports opened and forwarded. is it still impossible? URiel Nope, it is not currently possible. * behind a NAT for SIP does not work because the * real IP address is placed in the SDP information, therefore the 'outside' phone can not send the media stream to *. See my answers over the last week for the more details and possible work arounds. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
A plain wireless dlink dsl router. Stephen Varga wrote: On Thu, 2003-09-25 at 10:42, Michael Koehler wrote: Sorry, but my * is behind NAT and i have no problems with SIP, and it even works with NAT to NAT and without forwarding ports or similar effords. Michael What kinda box/device is doing the NAT? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
Looks interesting, got retail prices? Dave Cotton wrote: On Thu, 2003-09-25 at 18:54, Michael Koehler wrote: A plain wireless dlink dsl router. I'm testing one of these http://www.intertex.se and my * is behind it.
Re: [Asterisk-Users] SIP / GrandStream Configuration
Europe (Germany) and US (Calif.) Dave Cotton wrote: On Thu, 2003-09-25 at 19:10, Michael Koehler wrote: Looks interesting, got retail prices? Where are you in the world?
Re: [Asterisk-Users] ERROR MESSAGE
This means that your Request is not confirmed after N tries. This could happen for example) when your NAT is not working with your VoIP provider. Other reasons possible (e.g. no internet connection) listas iPfone wrote: Hi I have that error messages, what does it mean? *CLI WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Request) WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Request) miklos ___ 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
Re: [Asterisk-Users] SIP / GrandStream Configuration
It is not a feature of the router, it is the way SIP is handled with nikotel.com I recently wrote that i'm using just a plain router with my natted asterisk because "Stephen Varga" wrote that SIP behind NAT (in relation to asterisk) is impossible. It is possible because i'm using asterisk this way. There is also nothing special to setup with the router for nikotel and NAT, except you have a firewall and need straight rules, then you may use port forwarding. Michael Stephen Varga wrote: On Thu, 2003-09-25 at 12:54, Michael Koehler wrote: A plain wireless dlink dsl router. Do you know the model number and the software version? I am trying to understand how it is making the appropriate adjustments to allow the connection to work. Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extract header(s) of SIP signalling messages
googled: yes, asked #asterisk: yes.. I need to extract headers of the SIP call signalling protocol on outbound calls (especially call setup response message/ 200 OK). Example: phone with extension 33 place a call to the number 12345 (or sip:[EMAIL PROTECTED]:5060) via SIP. The SIP proxy then sends a header within a call setup response which is important for further CDR processing. I appreciate any idea. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream Source in the EU?
Nikotel is shipping from US and Germany. Germany to France will take 3-4 days in shipping. Louis-David Mitterrand wrote: On Mon, Sep 15, 2003 at 10:27:15PM +0200, Dave Cotton wrote: On Mon, 2003-09-15 at 22:11, Tom (UnitedLayer) wrote: Anyone have a good source for BT-101 phones? Yes. But it may not work for you because I've no idea on which of the 5 continents you are. I am looking for Grandstream phones (BT-102) in France, EU. Importing them from the US seems really a waste of money since one would pay double duty (.cn(?) - .us and .us - .fr). Is there a GS distributor in the EU or is it possible to order them direct from the manufacturer? Thanks for any info,
Re: [Asterisk-Users] Re: Grandstream Source in the EU?
Sorry, did not catch up that you want to have a bt-102, we have just bt-101 in stock currently which have only one network connector. Anyway, bt-101 is for free, you just have to open an account at nikotel and charge your account with 119 euro (inkl. taxes) to receive the bt-101. Shipping is 10 eur to france. Bests, Michael Michael Koehler wrote: Nikotel is shipping from US and Germany. Germany to France will take 3-4 days in shipping. Louis-David Mitterrand wrote: On Mon, Sep 15, 2003 at 10:27:15PM +0200, Dave Cotton wrote: On Mon, 2003-09-15 at 22:11, Tom (UnitedLayer) wrote: Anyone have a good source for BT-101 phones? Yes. But it may not work for you because I've no idea on which of the 5 continents you are. I am looking for Grandstream phones (BT-102) in France, EU. Importing them from the US seems really a waste of money since one would pay double duty (.cn(?) - .us and .us - .fr). Is there a GS distributor in the EU or is it possible to order them direct from the manufacturer? Thanks for any info,
Re: [Asterisk-Users] Grandstream Source?
You get a Budgetone for free at Nikotel if you charge your account there with 100 bucks. The nikotel service works with *, even behind nat Tom (UnitedLayer) wrote: Anyone have a good source for BT-101 phones? I had a lead on some, but they've not materialized. I'm also interested in the ATA-286 (HandyTone) units as well. This is for my personal Asterisk/INOC-DBA setup, that has yet to materialize heh. --- Tom Sparks ___ 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] chan_sip, nat, no media before 200 OK
My * box is behind NAT and works well. The only issue is that when i make a outbound call i get no media with 183 Session progress w/sdp before the session is setted up with sipmsg 200 OK/ACK The pstn gateway is a AS5300 and it runs the newest image. Media address detection is enabled on the AS and when i'm passing the asterisk by and use the proxy instead of the asterisk then it works fine. I tinkered with the Dial() application flags already w/o positive results. 2 Questions: Is there anything else i have to set up at sip.conf. Are SIP call flowcharts available for chan_sip ? michael parts of sip.conf: == [koehlerisk] username=koehlerisk fromuser=koehlerisk type=friend context=incoming host=calamar0.nikotel.com canreinvite=yes mailbox=10 nat=1 parts of extensions.conf: [standard] exten = _001X.,1,Dial(SIP/[EMAIL PROTECTED],60,Tt) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA v2.16 Register Update problem
i assume we talking about ata and sip. 2.14 and 2.15 does not have this bug. I have the same problem w/ fw v2.16 when getting anything else then "100 trying" or "200 OK" on the register callleg, which either results in stopping to register (i power cycle the ata or go to the web config and update once then) or registering with a malformed request uri "REGISTER sip:: SIP/2.0" where host and port disapeared. i suggest to upgrade the fw Michael Dan Fernandez wrote: Ive got an ATA with v2.16 configured as per JTodds guide having problemsre-registering. The only thing particular to this ATA(Ive got several all configured the same way but one of them have problems) is that is on a public IP. I have commented NAT=1 and canreinvite=no on sip.conf The ATA registers fine the first time but on the following registration updates, * gives me the the warning: "Registration fail for ." and it continues trying to register unsuccesfully over and over. If I do sip show peers, the ATA is not there, of course. Looking at the archives Ive found that on the new version (2.16.1) they have fixed a bug which is somewhat related: CSCeb17953 The Cisco ATA stops the registration process if it receives an unexpected response to a REGISTER request. HAs anyone had this problem before with 2.16? Woudl 2.16.1 fix this ? Thanks in advance
