Re: [Asterisk-Users] Working example of switch?
Ian Blenke wrote: Does anyone have a working example of how to use the switch directive to peer two Asterisk PBXes? [provider] switch = IAX2/[EMAIL PROTECTED]/context Then include = provider in the appropriate context On the local host you need a type=peer for your 'peer' Then on the remote host you need a type=user for 'user' and the extensions in [context]. Good luck, find me on IRC if u can't figure out WTF i'm talkin about (its late) Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone Defective Units
James, The remote NTP problem is fixed in the current beta firmware that I have been testing for GS.. Hopefully it will be released soon.. Later I have not had any problem at all with the 10 I have. They sound good and work well. The only problem I ever had was a problem with remote ntp servers. Andres wrote: Hi, I would like to know if others have experienced a high percentage of Budgetone defective units. We purchased 4 to test with our Asterisk. One was DOA and the other died after 3 days. So far we have a 50% failure. This does not look good. Let me know if its just us or if it is widespread. On the other hand we have purchased about 50 ATA186s and none of them have failed. Thanks, Andres ___ 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 -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pager support
I have a request from a customer to provide several levels of paging when someone leaves a VM. On other PBX's VM systems it is common for the system to 'dial out' to a pager wait a few seconds and then send a DTMF sequence that is usually the mailbox number, or sometimes it will call a remote phone number and ask for a password and let them listen to their VM message. I have set something up using cron and shell scripts with app_queuecall but I was looking for a more elegant solution. Does anyone have something like this already coded. I will gladly post my scripts if anyone asks. I have seem some discussion of an application called app_hasvoicemail has anyone seem this?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ATAs
Roger De Salis wrote: Immediately strides to ATA, rips off cover... woohoo, EEPROM is socketed well maybe I'll just copy the contents of a working ATA into the programmer, and reflash the locked one, taking care to change the MAC address and serial number.. Keep us posted, please. Really our karmic outrage should be vented equally at Cisco and at Vonage, which is the progenitor of the password locking scheme. Vonage will get extra eons in Hades for their misleading advertising, which until recently gave their customers the illusion that they were being given the ATA186 as part of their purchasing the service. IMO anyone who purposely designs a scheme to render perfectly good hardware useless deserves to see their user base transfer en masse to NuFone. . . B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 21 August 2003 07:02, Manoj K Gupta wrote: you can also try setting AMAflag=billing in oh323.conf if it helps. Already tried that. Didn't make a difference. I'll test again with the latest oh323 (0.5.5). - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/RdwD2TEAILET3McRAnuhAKChXRMQE6Z+Gc4WDjv734JrIIHGLwCfQqeU 4BqTSpgKHlAbFZ/XVKNgWIY= =deYU -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sox and wav to gsm conversion quality issue
Hi, I have some voice prompts in WAV format (PCM, 44KHz, 16bit, mono, bitrate 705kbps) and I want to convert them in gsm format. Using : sox file.wav file.gsm The result is a gsm sound file which when is played the speed is very very low (something like 5-6 times slower). Converting first to a 8KHz, 8bit mono wave file, the final gsm file is correct, but the quality is far than acceptable. The original file is at a very high quality. Which is the procedure to obtain the highest quality possible from my original wav file? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Slowly get it ... Hardware
Hello, i just got asterisk up and working (without alsa). Looks really promising. Now i have some questions regarding hardware I have the following setup PSTN --- serveral PBX (Asterisk) --- digital phones PRI (S2M) Ports analog phones VoIP On the PSTN side i could use Eicon/DIVA PRI 30M cards. Would a Digium T100P or E100P also work in germany ? For conferencing it looks as I would need one Zaptel card. What is recommended in such a setup (if I can't use Digium equipment on the PSTN side) ? What is needed to connect analog and digital devices ? For analog devices it looks as I need T1 interfaces and channel banks. Is that ok ? What hardware is recommended ? And how can I connect ISDN (digital) equipment ? Is there a way to do that ? Thanks Peter -- dadi-linux www.dadi-linux.de Peter Eckhardt Fon: +49 6071 951256 Weberstr. 36BFax: +49 6071 951257 64846 Groß-Zimmern [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sox and wav to gsm conversion quality issue
Dan wrote: Hi, I have some voice prompts in WAV format (PCM, 44KHz, 16bit, mono, bitrate 705kbps) and I want to convert them in gsm format. Using : sox file.wav file.gsm The result is a gsm sound file which when is played the speed is very very low (something like 5-6 times slower). Converting first to a 8KHz, 8bit mono wave file, the final gsm file is correct, but the quality is far than acceptable. The original file is at a very high quality. Which is the procedure to obtain the highest quality possible from my original wav file? Thanks, Dan Sox can convert the bitrate and the format at the same time, which in my experience produces pretty good results. I posted the process that I use to the list a few weeks ago: http://lists.digium.com/pipermail/asterisk-users/2003-July/016399.html Regards, -- Jamie Neil [EMAIL PROTECTED] Versado I.T. Services Ltd. http://versado.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sox and wav to gsm conversion quality issue
Sox can convert the bitrate and the format at the same time, which in my experience produces pretty good results. I posted the process that I use to the list a few weeks ago: http://lists.digium.com/pipermail/asterisk-users/2003-July/016399.html Hi Jamie, If I try like in your example and first resample the original at 8KHz, the quality of the final sound is unacceptable. BR, Dan Regards, -- Jamie Neil [EMAIL PROTECTED] Versado I.T. Services Ltd. http://versado.net/ ___ 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] sox and wav to gsm conversion quality issue
Ok...some more tests and the best results I get are with: sox file.wav -r 8000 file.gsm resample -ql Now the quality is identical with the original Asterisk prompts. Thanks for he hint. Dan - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 22, 2003 3:02 PM Subject: Re: [Asterisk-Users] sox and wav to gsm conversion quality issue Sox can convert the bitrate and the format at the same time, which in my experience produces pretty good results. I posted the process that I use to the list a few weeks ago: http://lists.digium.com/pipermail/asterisk-users/2003-July/016399.html Hi Jamie, If I try like in your example and first resample the original at 8KHz, the quality of the final sound is unacceptable. BR, Dan Regards, -- Jamie Neil [EMAIL PROTECTED] Versado I.T. Services Ltd. http://versado.net/ ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 19 August 2003 19:21, Michael Manousos wrote: Looks like mysql_log() is not actually getting called. Try a pure Zap bridged call or an IAX2 call (both of which I've tested) and see if it still doesn't work for you. Are you getting a log from cdr_csv in the log file (i.e. is CDR logging working at all)? Hmm... A zap bridged call created a CDR record. And a zap - oh323 call also inserts a CDR. Strange. What could cause oh323 initiated calls suddenly not generate CDRs? The CDRs are generated by Asterisk. It is not something that is controlled by the channel driver. Well, whatever was causing this seems to be gone with latest asterisk cvs and oh323 0.5.5. Thanks all. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/Rg6z2TEAILET3McRAqv/AJoC9lkKfPy1z/jAuwFPRT6IKbCjogCfSt0W lrCvxLvDj6fYiHZ4PUqWS5o= =P6LH -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pops
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 19 August 2003 19:18, Michael Manousos wrote: Could you provide some more details on the configuration and your system setup? Configuration of OpenH323 channel driver - Version: 0.5.5 Listening on address: x.x.x.x:1720 Gatekeeper used: No Gatekeeper FastStart/H245Tunnelling/H245inSetup: ON/ON/ON Supported format(s): G.711A Jitter buffer limits (min/max): 20-100 ms TCP port range: 1 - 2 UDP (RAS) port range: 1 - 2 UDP (RTP) port range: 1 - 2 IP Type-of-Service value: 16 User input mode: 2 Max number of inbound H.323 calls: 1000 Max number of outbound H.323 calls: 1000 Box: Intel Xeon 2.4GHz, 533 FSB, 1GB RAM. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/Rg9f2TEAILET3McRAuPXAJ9TZaJ4eaSJAqktdqtJQl2FRgodwwCdGMpv Wz61BYMBbdOPuRuY9K0vJkU= =nsIW -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to change Contact info?
I wanted to know if it is possible to change the contact info so it would be [EMAIL PROTECTED] instead that [EMAIL PROTECTED] If this is possible could you please give me the info. Thanks, Best regards, Yehiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best SIP phone?
Hello, I would like to request opinions for: What is the best SIP phone? Please give a rating from 1 to 10 (10 being the best) for the following categories: Reliability Ease of deployment with Asterisk Sound quality Please be sure to supply the manufacturer's name and the phone's model number. If you have an opinion on this topic, I would like to hear it. -- Thanks, Tim Best is relative to how much you are wanting to spend per phone and what features you are after.. At the end of the day you can get a Grandstream Budgetone for $75, a Snom200 for $250 or a Cisco for $700.. They all work perfectly with Asterisk as phones to make phone calls and to hear the person on the other end.. All have good voice quality.. (Remember voice quality is ver much dependent on which codec you choose to use.) If however you want downloadable ringtones and XML interfaces to the phones diaplay then you will have no choice but to fork over the cash for the Cisco.. Not sure is that helps you at all.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE:911, networks of * servers, etc. (was: VOIP Dialtone?)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: 21 August 2003 21:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE:911, networks of * servers, etc. (was: VOIP Dialtone?) Yes, I'm familiar with the E911 platforms and their requirements to some degree. The trick is that the people running Asterisk PBX systems have no visibility into SS7, and that is an unreasonable expectation, so some other out-of-band method for moving caller location to the PSAP is required. As far as geographic location tracking is concerned: that is the user's problem. If they don't have the correct information in their device, then they're SOL. There is _no way_ to develop lat/lon/alt coordinates from an IP address, despite what any .com flash-in-the-pan company says they can do with their clever databases. Thus, the PBX/switch provider will have to enforce their own database of device-to-geographic-coordinates. (As mentioned, maybe a SIP header is a reasonable thing to use for the UA to relay this data to the proxy.) I am not concerned so much about the ability of the devices to send their data to the proxy: I am VERY concerned about how the proxy then looks up the appropriate PSAP, and then relays the data for the call to that PSAP. JT 911 through the phone system is tricky business. e911 which is the automated process of handing the address to the 911 center uses the SS7 database to do it's work (the database is created when the LEC runs physical lines to locations not by people filling anything out). Cell phone service providers have the simuliar problems as VoIP service providers are facing are realizing with call forwarding and call following it will get worse.. Congress has mandated that the cell phone industry make it possible to track a cell phone users within 300yards via cell sites and triangulation. By 2005 every cell phone will be required to have a GPS and send GPS information to the 911 system when they call 911. If you want more information on e911 try http://www.fcc.gov/911/enhanced/ . As the cell phone industry grows there will be a need for a national 911 call routing center. I bet it won't be free. Original Message: - From: John Todd [EMAIL PROTECTED] Date: Thu, 21 Aug 2003 01:32:24 -0700 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 911, networks of * servers, etc. (was: VOIP Dialtone?) OK, that VOIP dialtone? thread was getting really out of hand, so I'll condense my answers into one big ugly message: 1) 911 service. Yes, that is one of three reasons to keep your PSTN line. The other two reasons are: Inbound calls from local callers still should work on a POTS line, for now. You can't find VOIP providers in most area codes, so you'll most likely need to have a local number that finds it's way to you for local tasks. Secondly, the Internet is not as reliable as the phone system. Sorry, folks, it just works that way right now despite what your network engineer might tell you. That's not to say it's unreliable, but those last two nines are very expensive... Besides, any good network engineer will tell you that you should have multiple paths for your IP connectivity. With few exceptions, most homes do not have multipath connectivity. (note: businesses may in fact have better uptime on their IP network than their phone network, if they have competent engineers and a reasonable budget.) 1.5) There are reasonable technical solutions to this problem, but for the life of me I can't figure out why the 911 centers haven't gotten their act together and solved this. There are two halves to this problem: What PSAP do I call? (and what phone number) and How do I get my location data to the PSAP once I call them? C'mon, this is not difficult. The first question can be answered trivially: there _must_ be a database of address-to-PSAP mappings. Any PBX administrator (or SIP phone owner, for that matter) should be able to figure out their address. Methods for associating the PSAP number with the phone are numerous, and trivially implemented - if people don't keep their address information updated, they're SOL (though you can remind them in an automated fashion to keep it updated - just forbid them from using the service unless they verify the address every month or so.) The second question is more difficult, but certainly possible. There may be kludge ways of doing it, and there should be more elegant ways of doing it. A SIP header with lat/lon/alt data that gets sent from the UA only on 911 (or other programmable string) calls might be reasonably elegant... maybe. But that only gets the data to the SIP proxy. That doesn't solve the issue of how you get that data from the SIP proxy to the PSAP, which at some point will be almost certainly through a PSTN connection... ADSI FSK, maybe? Ugly, and PSAPs would not want to invest in equipment. A national
Re: [Asterisk-Users] sox and wav to gsm conversion quality issue
sox infile.wav -r 8000 -c 1 outfile.gsm On Fri, 22 Aug 2003, Dan wrote: Hi, I have some voice prompts in WAV format (PCM, 44KHz, 16bit, mono, bitrate 705kbps) and I want to convert them in gsm format. Using : sox file.wav file.gsm The result is a gsm sound file which when is played the speed is very very low (something like 5-6 times slower). Converting first to a 8KHz, 8bit mono wave file, the final gsm file is correct, but the quality is far than acceptable. The original file is at a very high quality. Which is the procedure to obtain the highest quality possible from my original wav file? Thanks, Dan ___ 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] Structured release, Maillists
Is there any hope for a 0.5.0 release on the horizon? I would also like to see a more structured release program. It's kind of scary to tell people that they should just use the latest CVS code. 0.5.0 is coming up soon, but I would like to resolve relevant CRASH and MAJOR issues in the bug tracker first. After 0.5.0, I believe we will be on a path for Asterisk 1.0. In general, no instrusive new features will be permitted, and we will need people to test, but more on that when we get there. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need a trick to generate calls
Hi- I wonder if anyone knows of a way that I can have some event (like an incoming phone call or other stimulus) generate a lot of outgoing calls on an asterisk system. I want to use one asterisk system to load-test another, and this would be necessary. I know that I can have one incoming call generate an outgoing call (or even several calls until one is answered), but I want to generate lots (dozens) of outgoing calls that continue in parallel even if one is answered. Thanks Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need a trick to generate calls
Scott, you can write a file into /var/spool/asterisk/outgoing/ to trigger a call. Some parameters I know are: Channel: Zap/g1/123412341234 MaxRetries: 0 RetryTime: 1 WaitTime: 1 Context: out-context Extension: 432142314231 Priority: 1 Callerid: 11 Thilo On Fri, 2003-08-22 at 16:08, Scott Stingel wrote: Hi- I wonder if anyone knows of a way that I can have some event (like an incoming phone call or other stimulus) generate a lot of outgoing calls on an asterisk system. I want to use one asterisk system to load-test another, and this would be necessary. I know that I can have one incoming call generate an outgoing call (or even several calls until one is answered), but I want to generate lots (dozens) of outgoing calls that continue in parallel even if one is answered. Thanks Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- [netzquadrat] GmbH | Thilo Salmon Ronsdorfer Str. 74 | Fon: +49 211 302033 12 40233 Duesseldorf| Fax: +49 211 302033 22 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to change Contact info?
try adding fromdomain=externalip to your sip entries Martin On Fri, 22 Aug 2003, Yehiel Samson wrote: I wanted to know if it is possible to change the contact info so it would be [EMAIL PROTECTED] instead that [EMAIL PROTECTED] If this is possible could you please give me the info. Thanks, Best regards, Yehiel ___ 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] cdr_csv actual duaration
Hi all. I have some question about cdr_csv. We want to have some IVR system for routing incoming calls to multiple directions. So when user make a call, he must put his PIN and phone number. Then asterisk will Dial this call to appropriate destination. Billing system deal with cdr_csv records. But the Billable seconds field - all time of the call, including the IVR part (that haven't be billed). And I don't know how to save the actual call duaration into the cdr_csv record. Maybe cdr_mysql doesn't have this problem? Any ideas? Thanks. __ www.newmail.ru -- - . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF tones not long enough on out going calls
DTMF tones are not long enough on out going calls, when I'm using either info or rfc2833. Does anyone know if the tone length value is in rtp.c or chan_sip.c ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + SNOM + Pound and star keys
At 06:43 AM 8/22/2003 +, you wrote: How are people handling call transfer with SNOM phones? We are okay with the # transfer workaround, but I worry about how that will work with other systems that expect me to be able to press # to return to the previous menu or similar. Thanks, --Ernest Whats wrong with using the SIP transfer built into the SNOM? That's pretty much my question :) I've heard that it doesn't work well, and testing here confirms that. Transferring someone to and extension just disconnects both ends. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF tones not long enough on out going calls
Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !? -Original Message- From: James Sizemore [mailto:[EMAIL PROTECTED] Sent: 22 August 2003 17:33 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] DTMF tones not long enough on out going calls DTMF tones are not long enough on out going calls, when I'm using either info or rfc2833. Does anyone know if the tone length value is in rtp.c or chan_sip.c ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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
Re: [Asterisk-Users] Question on setting up MeetMe conference bridge
I was able to connect two H323 clients in a conference call with only one X100P card. ztdummy did not compile correctly so I'm only using zaptel.o and wsfxo.o. The conferencing worked without any special setup on my part so I can't say that it's anything that I've done. - Original Message - From: Lee Goodman To: [EMAIL PROTECTED] Sent: Thursday, August 21, 2003 10:06 AM Subject: [Asterisk-Users] Question on setting up MeetMe conference bridge So I setup the MeetMe application in Asterisk Assigned an extension to it. When one of my SIP phone dials the conference extension, they get a message "you are the first one in the conference", so far so good. When the 2nd SIP phone dials the conference extension, they get a busy signal Now I know that you have to have a Zapta device to enable conference application. I have an X100P (1 port FXO card). So, do you have to have mulitple Zapata devices to enable multiple users in a conference? Do I have to enable the zdummy (dummy zapata code) to run a multiuser conference? Thanks Lee Goodman PS, if someone could send an example configuration, that would be great
Re: [Asterisk-Users] Asterisk + SNOM + Pound and star keys
I have the some problem. I cannot use # on my cisco ATA and also not on X-Lite phone. When i try to transfer on soft phone with trasfer button it just disconnects both ends. Bart - Original Message - From: Ernest W. Lessenger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 22, 2003 1:01 PM Subject: Re: [Asterisk-Users] Asterisk + SNOM + Pound and star keys At 06:43 AM 8/22/2003 +, you wrote: How are people handling call transfer with SNOM phones? We are okay with the # transfer workaround, but I worry about how that will work with other systems that expect me to be able to press # to return to the previous menu or similar. Thanks, --Ernest Whats wrong with using the SIP transfer built into the SNOM? That's pretty much my question :) I've heard that it doesn't work well, and testing here confirms that. Transferring someone to and extension just disconnects both ends. --Ernest ___ 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] DTMF tones not long enough on out going call s
Have someone using a SIP device with RFC2833 signaling call you, now have the press and hold down one of the dialing keys. You'll hear a short tone then nothing. On Fri, 2003-08-22 at 11:05, Low, Adam wrote: Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !? -Original Message- From: James Sizemore [mailto:[EMAIL PROTECTED] Sent: 22 August 2003 17:33 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] DTMF tones not long enough on out going calls DTMF tones are not long enough on out going calls, when I'm using either info or rfc2833. Does anyone know if the tone length value is in rtp.c or chan_sip.c ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI CallerID problem
Hi Michael, I've run into the same problem. My installation is using PRI
and I've tried setting Caller ID the same way you have. Same results, it
works fine when we call out from a SIP device but rerouting an inbound zap
channel out to another has been unresolvable.
Let me know if you find a solution - my customers are frustrated by the
loss of Caller ID on forwarded calls.
Steve Bourg
On Wed, 20 Aug 2003, Michael Rose wrote:
Greetings all..
We have an inbound/outbound PRI installed and terminated on a T400P
Digium Quad T1 card. Were seeing an odd problem when sending
$CALLERIDNUM when calls from the PRI are forwarded back out to the PSTN
over the PRI. The $CALLERIDNUM is not being sent out along with the
call. Its sending the phone number of the PRI itself, rather than the
$CALLERIDNUM information.
Yes, we can send CID info to our PRI provider. If we make a call with
our Cisco 7960, we can send any phone number we enter into
SetCallerID(). However, if we use SetCallerID(${CALLERIDNUM}) it wont
forward the CID number.
[mydid]
exten = 5558384810,1,SetCallerID(${CALLERIDNUM})
exten =
5558384810,2,Dial(Zap/g1/15553456131SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED],20)
exten = 2068384810,3,Congestion
Now when I call my DID, 5558384810 from my land line, Asterisk takes the
CID number and sends it to my IP phones (Cisco 7960) which I can see on
the display of the IP Phone. However, the CID number doesnt get sent to
the cell phone.
Below is the context we use to make outgoing calls from our Cisco
7960s. This works fine with our PRI. When I call using the following,
my cell phone shows the incoming call coming from 2065551212. This tells
me that our PRI vendor is allowing us to send CID info.
[dialout-pri]
exten = _1NX,1,SetCallerID(2065551212)
exten = _1NX,2,Dial(Zap/g1/${EXTEN},100,T)
exten = _1NX,3,Congestion
exten = _1NX,4,Hangup
Heres what Asterisk shows, when the call is coming in.
-- Executing SetCallerID(Zap/23-1, 5557209085) in new stack
-- Executing Dial(Zap/23-1,
Zap/g1/15557206131SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED]|20) in new
stack
-- Called g1/15557206131
-- Called [EMAIL PROTECTED]
-- Called [EMAIL PROTECTED]
-- Accepting call from '5557209085' to '5558384810' on channel 23,
span 1
-- SIP/69.28.200.84-4698 is ringing
-- SIP/10.0.1.2-6695 is ringing
-- Zap/1-1 is ringing
Thanks in advance.
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Re: [Asterisk-Users] RTP header compression?
cRTP (compressed RTP, and its official acronym has a small 'c') is done at the router level, so it's transparent to the user agent. AFAIK, there's no open implementation of cRTP. Cisco switches support it. Another approach is ROHC (Robust Header Compression), it works on all IP packets. This one is more promising, it's also the standard for 3G phones. ROHC can be implemented at end-points like the 3G phones, not just routers. There's a Linux ROHC implementation for improving throughput on slow links like PPP. See http://rohc.sourceforge.net/ Hope this helps. Kevin K wrote: Uh oh. I think I may be looking at the wrong tool. My goal is to implement (in an open source software suite) an RTP/UDP/IP header compression algorithm that would save bandwidth used by voice traffic packets. So a 5ms G.711 packet that would otherwise be 98 bytes, could be reduced to 62 bytes when its RTP/UDP/IP header is compressed. This would require me to get into the RTP stack used to packetize the actual data (voice) packets. From what I am reading, * is not the right software for that kind of work because it don't really touch the voice packets themselves but rather enables the two endpoints to talk to negotiate and connect with each other (as well as utilize a number of other features - vmail, callerID, etc). Is this correct? Is what I am looking for an open-sourced SIP user agent? Or is it more than that? Any help would be great, as I appear to be confused. Thanks, Kevin Original Message Follows From: John Todd [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP header compression? Date: Wed, 20 Aug 2003 13:38:22 -0700 I sent this to the asterisk-dev by accident... Original Message Follows Hi all, I have a couple questions about RTP header compression with Asterisk: 1) Has this been implemented before or is it included in the Asterisk package? 2) If the answer to (1) is no, is there an RTP stack that this can be logically implemented into? Where would that be? Thanks, Kevin As I mentioned before in a reply to this: 1) No. 2) Take a look at the source code; you should be able to find the rtp sections in it readily enough. 3) If you are asking about RTP compression between Asterisk servers, look at IAX2 and trunking mode. Questions 1 and 2 become less meaningful. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ The new MSN 8: smart spam protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ 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] CVS Question
In order to get the version to update, you can run make update in the asterisk directory, instead of doing a cvs update. Jared Smith On Fri, 2003-08-22 at 20:01, Andres wrote: When I checkout the latest asterisk version, then do a make clean and make install, shouldn't the show version command tell me the new version date? I have done several updates and the version displayed is still from 2 months ago. Thanks. Andres. ___ 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] Game time is over gang
Looks like it about to hit the fan I think we missed the fidonet of phones opportunity http://news.com.com/2100-1037_3-5066652.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone Defective Units
-= On Fri, 22 Aug 2003 06:49:53 +, WipeOut . [EMAIL PROTECTED] said: The remote NTP problem is fixed in the current beta firmware that I have been testing for GS.. Hopefully it will be released soon.. I'm going 'round and 'round with the GS support guys. I've tried three beta versions of the firmware and still don't have a fix for what appears to be a serious problem. If I use a DNS server inside my LAN (within the subnet) while my router is not at that same address, the phones crash at boot time. In other cases, they will ARP crazy things, like remote servers or 0.0.0.0. I could be wrong about the specifics, but there seems to be certain sets of network addresses that they handle poorly. I've sent them a number of packet traces and some repeatable scenarios, but each firmware update seems to fix one and break others. At this point, I don't have a lot of confidence in their TCP/IP stack. That said, none of the three units (or their power supplies) have failed on me. :) For best results, make sure the DHCP server and Router are the same machine, and that the DNS server is outside your subnet. -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS Question
It worked. Thanks! On Friday 22 August 2003 21:02, Jared Smith wrote: In order to get the version to update, you can run make update in the asterisk directory, instead of doing a cvs update. Jared Smith On Fri, 2003-08-22 at 20:01, Andres wrote: When I checkout the latest asterisk version, then do a make clean and make install, shouldn't the show version command tell me the new version date? I have done several updates and the version displayed is still from 2 months ago. Thanks. Andres. ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pardon the newbie question
I hope you all will excuse my ignorance I just setup Asterisk for the first time and am trying to find everything I need to change, can someone point me to an article or document that would explain how to do a generic voicemail setup via a voice modem, if this can't be done I hope you'll be gentle :) Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Game time is over gang
Looks like it about to hit the fan I think we missed the fidonet of phones opportunity http://news.com.com/2100-1037_3-5066652.html You've got it backwards. This is precisely the reason we should implement it. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Game time is over gang
I don't put much faith in that lasting too long. They don't regulate email or can't for that matter. They don't regulate streaming video... Its a battle they will loose. Its data packets. If they get away with regulations on this.. whats next regulation of FTP or HTTP for that matter? And as I finish this email I see that jtodd pointed out this is the exact reason we would be doing this. I totally agree. Mark said he could map it thru iaxtel. HINT HINT if iaxtel would work for me. bkw On Fri, 22 Aug 2003, Bruce Ferrell wrote: Looks like it about to hit the fan I think we missed the fidonet of phones opportunity http://news.com.com/2100-1037_3-5066652.html ___ 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] Grandstream Budgetone Defective Units
Out of the 8 test Grandstreams we purchased in May- 3 of them are now totally dead. It looks like power supply problems... 2 of the phones have broken buttons... And every time I plug them into the network- half of them cause the network to crash. It seems that in the network we are testing on (a Dutch university) does not agree with the Grandstream phones. When plugged into the network, they seem to grab the MAC address from the local router and us it is their own. This produces total chaos- local pc's get confused and the network stops working. Not good. I told GS about this problem a few months ago- they initially blamed the problem on the network. So far, there has been no satisfactory solution. It seems obvious to me that the GS is not ready for prime time... -GSR - Original Message - From: Steve Haehnichen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 23, 2003 5:18 AM Subject: Re: [Asterisk-Users] Grandstream Budgetone Defective Units -= On Fri, 22 Aug 2003 06:49:53 +, WipeOut . [EMAIL PROTECTED] said: The remote NTP problem is fixed in the current beta firmware that I have been testing for GS.. Hopefully it will be released soon.. I'm going 'round and 'round with the GS support guys. I've tried three beta versions of the firmware and still don't have a fix for what appears to be a serious problem. If I use a DNS server inside my LAN (within the subnet) while my router is not at that same address, the phones crash at boot time. In other cases, they will ARP crazy things, like remote servers or 0.0.0.0. I could be wrong about the specifics, but there seems to be certain sets of network addresses that they handle poorly. I've sent them a number of packet traces and some repeatable scenarios, but each firmware update seems to fix one and break others. At this point, I don't have a lot of confidence in their TCP/IP stack. That said, none of the three units (or their power supplies) have failed on me. :) For best results, make sure the DHCP server and Router are the same machine, and that the DNS server is outside your subnet. -Steve ___ 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] Intresting.. hrm
And it runs linux. http://www.zip4x4.com/ZIP4x4.htm Anyone seen one? bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intresting.. hrm
The real question is: How much? On Fri, 2003-08-22 at 23:44, Brian West wrote: And it runs linux. http://www.zip4x4.com/ZIP4x4.htm -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intresting.. hrm
Linuxdevices says $400 http://www.linuxdevices.com/articles/AT9406437906.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Saturday, August 23, 2003 1:33 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Intresting.. hrm The real question is: How much? On Fri, 2003-08-22 at 23:44, Brian West wrote: And it runs linux. http://www.zip4x4.com/ZIP4x4.htm -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ 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
