[Asterisk-Users] Re: grandstream budgetone-100 updates
dean collins wrote: Im using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.2025 Jan 18:25 Error Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg.txt Octet, Send192.168.16.2025 Jan 18:25 Error Can anyone tell me why these fail each time? Probably because the files are not in your TFTP root. This is probably because you are not using these files to autoconfigure your phones. Stephen R. Besch ___ 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] Re: four wildcards in a single pc
Jim Van Meggelen wrote: OK, look, you _might_ be able to free up enough IRQs on a PIC-based motherboard -- if you disable the serial ports, mouse, parallel port and USB. It's not recommended, but it's theoretically possible. And if you have a MoBo that is APIC-compliant, you should be able to have all the IRQs you can handle, so lack of IRQs doesn't need to be an issue (make sure you have a BIOS and chipset that's up to the task). BUT . . . Getting dedicated IRQs for the cards is a minor problem compared to what happens when you have four cards hammering away mercilessly at the chipset and CPU of your motherboard; 1000 IRQs per second, per card. Nobody's really sure what's wrong, but it causes problems for pretty nearly everyone. What everyone here is saying is that we're all pretty sure you're gonna run into problems; problems that could easily be avoided by avoiding the whole TDM400 mess in the first place. What am I missing? The Digium cards are unique in the world of telephony, because instead of having an expensive DSP chip on board, they use the CPU to provide this functionality. The challenge comes from the fact that voice is intolerant of delay. In order to ensure that the voice processing that goes on in the CPU is handled with no perceivable delay, the zaptel cards have to establish a kind of pseudo-synchronous clocking with the CPU. Unfortunately, the signalling bus on a PC isn't synchronous, at least not in that way. The clock that the zaptel cards use is the IRQ of the card, literally requesting the CPU interrupt what it's doing and pay attention to it 1000 times per second, regardless of what it's doing. You are proposing the use of FOUR of these cards. Since this has caused trouble for nearly everyone who has tried it, everyone is suggesting that you might want to give the matter some careful thought. There _are_ less painful ways. Cheers, Jim. Not so much to raise an argument as to make a few comments. Please remember that the following is mostly a synthesis based on comments in this thread. They are made in the interest of clarification and are subject to additional comments, corrections, flames, etc. 1) This is not to minimize the problem, but 1000 interrupts per second is quite a few, but not an overwhelming amount. Keep in mind that an unbuffered serial card (and there are more than a few of these out there) working at 19.2 Kbaud will rack up 1900 interrupts per second and the CPU doesn't even sweat. Even an old pig CPU wasn't much strained by 19.2 Kbaud. I used to run an interprocessor link between 2 PDP8's at 38.4 KBaud as a background task with a data terminal running at 19.2 KBaud, and it was hard to tell when data transfer's were going on. That computes to as many as 5700 interrupts per second. I suspect that something else is going on here. 2) While it is hard to estimate directly, if the driver is properly designed, the number of interrupts should not scale linearly. One drive should handle all cards, and part of the time there will be more than one card needing servicing on an interrupt. If the driver does not test for this, then it should. 3) No card is truly unbuffered. That would make the interrupt service headroom nearly zero. You would have to get to the card in the time it takes to transfer a single bit, that's only 15.6 microseconds at 64K. One must use at least a single buffer. From the comments made earlier in this thread, clearly the Digium card is using at least an 8-deep buffer. In fact, I suspect that the buffer is a standard 16-byte FIFO with the threshold set at 8 bytes. What this means is that it would be possible to handle all 4 cards in a single interrupt, depending of course on the design of the buffer, by always emptying all 4 card's buffers on any interrupt. In fact, only one card (i.e., the master) should even have interrupts enabled! 4) All of the preceding notwithstanding, I suspect that the real issue has nothing (or little) to do with interrupt load, but, given that the card uses CPU cycles rather than a DSP, the problem is more likley CPU overload from data handling, which in turn, causes missed interrupts. The point here is that the card designers made the choice to use the CPU as a DSP, assuming that a server box+card was cheaper than a more expensive card. In otherwords, the combination was intended to be the telephony apparatus, not just the card. If you make that assumption, then those who are successfully using more than one card per CPU are getting a really good deal. The second, third ... card is essentially a freebie. Sincerely, Stephen R. Besch ___ 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: Problem with Grandstream bt100
Ken D'Ambrosio wrote: R A wrote: Then what do you think i have to do? i install a sniffer and the phone make an ARP request to 67.153.142.69. the phone is 192.168.0.160 and i set my pc to 192.168.0.161 and i can't ping the phone. Get a new phone. :( What he's saying -- and I agree with him -- is that, if you power-cycled the phone while the lights were flashing (which, if you read the docs, means it's checking TFTP for a firmware upgrade), and the phone actually started the upgrade... you could now have a piece of junk instead of a phone. That's always been the problem with flash memory and the like -- if there aren't enough safeguards, it's all too easy to kill something during an upgrade. ALWAYS makes me nervous, for example, to upgrade the flash on my system's BIOS. Try a couple more power cycles, and maybe check the documentation for something I might be forgetting -- but it sounds like you have probably toasted the phone. Sorry! -Ken --- Holden Hao [EMAIL PROTECTED] wrote: Did you set a tftp server on your phone? If you did then most likely the phone was downloading the firmware when the leds were flashing. If you pulled the plug then you might have destroyed your firmware and the phone along with it. Holden Some revs of the GS firmware will wait forever for a reply from a time server or tftp server. If the address is configured and the phone is attempting to contact a server which doesn't respond, the lights never go out. If you can operate the menu, (you may need to cycle the handset off-hook and then on-hook first) set the TFTP and Time server addresses to 0.0.0.0 and restart the phone. If the phone now works, then you should be OK. Also, even if you unplugged the phone during a firmware update, you may still be able to get the phone operational if the bootloader code is intact (which is not that unlikely), but you will need to get a TFTP server located at the address that the phone is attempting to access. Hope this helps. Stephen R. Besch ___ 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: grandstream bt100 upgrade 1.0.5.18
Rodney Acosta Coya wrote: what is Beta firmware means? Rodney Simply that GS does not yet feel that the firmware is stable enough to release as recommended code. There must still be enough bugs in it to justify not releasing it. However, as Dave Cotton points out, there is a good possibility that the firmware is not the problem. It's just something that you must consider when a releas version of the firmware works and the beta version does not. Stephen R. Besch ___ 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: dont write me again
Steven Critchfield wrote: On Wed, 2004-12-01 at 10:24 -0500, Gregory Junker wrote: How about following the very easy-to-understand UNSUBSCRIBE procedure outlined at the bottom of every message from this list? (Oh gawd, I sound like Critchfield now :p ) Ohh noo, now you know that it doesn't take someone being mean or even mean spirited to get annoyed at the lack of effort some people exhibit. Exactly. Would those people who respond from the mailing list digest -PLEASE-PLEASE-PLEASE- do the following simple things: 1)Strip out the digest messages that have nothing to do with your reply. 2)Copy the appropriate subject line into your message subject before you send the message so that we can actually tell what you are writing about without having to search through your reply. Stephen R. Besch ___ 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: grandstream bt100 upgrade 1.0.5.18
Rodney Acosta Coya wrote: hi all i upgrade a bt100 phone and it can't resgister with asterisk Dec 1 13:25:49 NOTICE[1112980400]: chan_sip.c:7519 handle_request: Registration from 'sip:@172.16.4.249' failed for '172.16.4.226' is was working with the version 1.0.5.3 some bady now what is hapening? thanks in advance Well, 1.0.5.18 is Beta firmware. Stephen R. Besch ___ 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: Granstream BT100 - only partial success
George Burt wrote: snip [grandstream1] host=10.0.0.26 ; we have a static but private IP address canreinvite=yes; allow RTP voice traffic to bypass Asterisk snip IP Address: statically configured as: IP Address:10.0.0.26 Subnet Mask: 255.255.255.0 Default Router: 10.0.0.1 SIP Registration:Yes Comments: 1) You can't ask asterisk to register your phone if you have a fixed IP address specified as host= in sip.conf. Either the phone sends the address (i.e., host=dynamic), or you enter it as an IP address. It's OK to be fixed at the phone and dynamic in asterisk, but that isn't rational - just adds net traffic. Turn off the sip registration option on the phone. 2) Unless I am mistaken, you are not going to be able to use re-invites without NAT. It will work on your calls to analog phones handled by Asterisk and to other IP phones on the local network. However, as soon as you connect to an outbound/inbound service, the reinvite will fail and you will lose your media stream. 3) Don't know if it will make a difference, but I always set the router field to 0.0.0.0. There is no such thing as a valid router IP on a private network - they are not routable by design. I had quite an argument with Grandstream about this when I first purchased the phones. As a result, the firmware was modified to accept a null router entry for use with private IP ranges. Sincerely, Stephen R. Besch ___ 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: STUN and Asterisk? (Was: SER is a better NAT solution?)
Matthew Boehm wrote: STUN requires 2 NIC interfaces on the machine running the server right? And both interfaces need seperate public IP's right? 'And' the phones/ATA's need to support STUN right? I don't think the Cisco phones support STUN. Why 2 NICS. There should be no reason that you can't assign a public and a private IP to the same NIC. In fact, I'm doing exactly that on both my Windows and my Linux boxes. Stephen R. Besch ___ 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: Top posting
Gregory Junker wrote: I'll stop doing it when Walsh stops posting about it: http://www.faqs.org/rfcs/rfc1855.html (from the RFC) ...Don't wander off-topic, don't ramble and don't send mail or post messages solely to point out other people's errors in typing or spelling. These, more than any other behavior, mark you as an immature beginner. ___ 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 This all reminds me so much of Jonathan Swifts bit about the BigEndians and the LittleEndians (referring to which is the 'correct' end to open a soft boiled egg) in Gulliver's travels. Stephen R. Besch ___ 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: OT: Grandstream problems
How do you downgrade the Budgetone to 10Mb? I don't see anything on the configuration page to do that. Also the specs on the Budgetone say it is a 10Base-T port. You can't. It already is 10MB. It can't do 100MB. Stephen R. Besch ___ 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: Costum ring tones with BT10x
Paulo Adriano wrote: Hi, I just received some Grandstream phones to use with my Astrisk setup, and I´m wondering if there is some info around , rearding aditional ring tones. The default nes are not very nice imho. I need a strong ringing tone... Thanks in advance GS provides a custom ring tone generator for free - see their web site. Also there is at least one script available from the * mailing list. Stephen R. Besch ___ 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: Benjk's Question Why FXS
Wolf Paul wrote: How about a school strapped for cash, with around 60 POTS phones on hand and an almost free source of another 60? Versus a cost (here in Austria) of $99 for the cheapest VoIP phone (the cheapest Grandstream model). Of course that also means that FXS is only of interest if I can get it for under around $50-60/port -- if things cost more it becomes easier to argue early replacement of these POTS phones by IP phones. For that many analog phones, you probably want something like a channel bank(s), which can handle the both FXS and FXO lines and package them into T1's. These are pricey new, but if you have the flexibility - and many school financial officers frown on buying used or surplus equipment - you can get them on e-bay quite reasonably with a little effort. I was able to get my FXS/FXO cards (4 port) for about $25.00 (US) per analog line and a virtually new TSU-600 for $99.00. Stephen R. Besch ___ 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: Benjk's Question Why FXS
Andrew Kohlsmith wrote: IP Phones require massive rewiring of your network infrastructure -- throwing those phones with the built-in switches in the mix is just asking for trouble. -A. I agree - if you are have a hub based architecture. But not if you are using switches. And, sharing existing ports using a small 4-port switch (maybe 40 bucks or less) you don't even need much extra wiring. The packet traffic level from an IP phone is just not enough to be of any concern unless you are moving gigs of data simultaneously over the same shared port. Stephen R. Besch ___ 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: Grandstream Flashing (different issue)
Todd Routhier - Lightwave Technologies, LLC. wrote: OK, this is a different flashing issue than the one that's being talked about. I have a few of these phones (GrandStream 101) and when a voicemail is received the light on the LED starts blinking and the dial tine stutters, this is cool. BUT. How the do I get it to stop, I have had mine covered with paper for the last 2 days because the blinking LED panel is driving me nuts. I have received the messages in my email and looked at them on the web. I am thinking that I have to check them by calling into the Asterisk system and mark them as read or something in order for this to quite. Two problems, I don't know how to check them by phone just yet and I will likely never check them by phone. You absolutely must get the message file deleted from the mail store. It doesn't matter how you delete the message either. You can use the * phone interface or simply delete the files associated with the message. As soon as the files are gone from the INBOX, * sends a SIP command to the phone to turn off the MWI. As long as there are any active messages in the INBOX, the light stays on. If you would like to be a guinea pig for my VB program that allows you to manage your mail folders and messages from a Windows GUI, I'll send you a copy. The only caveat is that I haven't yet found a way to get perfect synchronization with file access to the mail store. The user needs to be careful not to modify the INBOX while * is taking a new message. For me this is usually not a problem, since I would never be not answering the phone when I am listening to/moving/deleting messages - as a result, * would never be writing a new message when I was using the VB program. The program requires the VB6 runtime (available from MS for free) and that you have SAMBA running on the * server. Sincerely, Stephen R. Besch ___ 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: cannot call Grandstream
Neil Cherry wrote: I never could get my time server to work with the GS. The following is an excerpt from my web page. I also use phones on a private subnet. Linux is RedHat. This works for me: The time service must be configured to allow the phones to request the time from your server, which must have at least one IP on the same private subnet as your phones, and can therefore can be reached without the need of a router (which won't pass any private subnet packets) and at least one IP on a public subnet so that it can reach a public time server. I run the time service on my Asterisk server. (You could set Asterisk up to be a master time server, but unless you absolutely must, I would not do this.) First, using the Date/Time setting panel in RedHat, specify a public time server either by name or IP. Once you have a valid time server defined, activate the NTP service on your server. You must activate the service first, since the RedHat applet removes any customizations that are added to the ntp.conf file when you activate the service from the Date/Time applet. (Is this stupid or what?) Once the service is running, add the following line to /etc/ntp.conf : restrict 192.168.xx.0 mask 255.255.255.0 notrust nomodify notrap Where xx is the common subnet of your phones and your server. Then, restart the time service from the services applet - NOT from the date/time applet! These settings configure the time service to be a time mirror. In other words, it merely accepts time settings from the public server you defined earlier, and passes them on to anyone that requests them, providing that they lie in the 192.168.xx subnet. After that, your phones should be able to retrieve the time/date from the server using any IP's that you have defined in the server that lie on your private network Stephen R. Besch ___ 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: cannot call Grandstream
Michael George wrote: On Wed, Oct 20, 2004 at 01:46:01PM -0400, Stephen R. Besch wrote: I have never been able to get the Grandstream to register reliably - with any version of the firmware. So you mean you don't use the Grandstreams, then? On the contrary, I use almost nothing but GS phones. As long as I avoid DHCP and don't use the periodic registration function, they are 100% reliable. I have 23 of them, and with the exception of reboots after power outages or when updating firmware, I have had virtually no serious problems with them. In the early days (that is 6 months back) the call waiting tone was obnoxiously loud. However that was fixed some time ago. Since then, I have been quite satisfied with them, especially considering the cost. Stephen R. Besch ___ 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: grandstream 102 flashing
[EMAIL PROTECTED] wrote: I checked using MS sbs network monitoring all it seems to be doing is asking for a ARP Rarp request to 67.153.142.69 C:\ping -a 67.153.142.69 Pinging ip67-153-142-69.z142-153-67.customer.algx.net [67.153.142.69] with 32 bytes of data: Reply from 67.153.142.69: bytes=32 time=140ms TTL=47 Don't know who algx.net is, but this may help. The other thing is it thinks it is ip address 192.168.1.160 but that isn't even part of my network. That is the default IP used when the phone ships. You might be able to use that to configure your phone using GSConfigure if you can find out what IP address the phone is using for TFTP and put a TFTP server on that IP. Even if the IP is public, you could still use it temporarily as long as your machine and the phone are isolated from the public network on a private switch/hub. Stephen R. Besch ___ 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: Voicemail and ast_data
Gunnar Schaller wrote: Hi, I have a problem with Voicemail. Asterisk 1.0.1 patched with ast_data connected to a mysql-server, mailboxes are in a mysql database. When I call to VoicemailMain to hear my messages it don't tell me the time the message was left. Only Message 1 and then the message. For testing I defined a mailbox in voicemail.conf. Hearing messages from this account in VoicemailMain tells me the time the message was left. For example Message one, received thuesday . and then the message. There are no special options I set. Can anybody give me a hint? Gunnar ___ 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 The audio files in the mail spool directory do not have the time/date embedded in the audio (.gsm or .wav). The time/date is in the .txt file that accompanies the message. When * plays the message, it gets the date/time from the .txt file and speaks it before playing the audio file. Stephen R. Besch ___ 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: cannot call Grandstream
Michael George wrote: I am having trouble with a Grandstream Budgetone 101. It's at firmware 1.0.5.10 and I'm running * 1.0.0. I have the phone getting a DHCP address and * expects it to register. When I reboot the phone it does register just fine. However, after a while * cannot contact the phone. I will call the phone and * will tell me: -- Called grandstream1 Oct 20 09:41:16 WARNING[98310]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) Looking in teh archives, it seems that that indicates that the registration is expired. I've got the phone set to 60m register intervals (and * acks that when the phone registers) but after the hour it doesn't re-register. I've also tried 15m and 2m register timeouts. I have Sip Registration and Unregister on Reboot both set to Yes on the phone. Register Expiration is 60. The phone is at 192.168.42.234 and * is as 192.168.1.3. Both internal but no NAT between them. And the initial registration works fine. I've searched through the mail list archives and tried all the suggestions I could find there, but the phone behaves the same: registration appears to be lost. Incidentally, I set the phone to a static IP (192.168.42.99) and also set * from host=dynamic to host=192.168.42.99 but * couldn't call the phone at all after that. (I did graceful restarts on * between the change). Can anyone see what I might be missing? I don't have the SIP UserID or Authenticate ID set to the phone's extension, but the SIP User ID is the same as the Authenticate ID which is the same as the context in *'s sip.conf. It doesn't seem that would have an effect, but I thought I'd mention it. Thanks! Michael, I have never been able to get the Grandstream to register reliably - with any version of the firmware. It sounds like in your test with the fixed IP, you left the registration option on the phone set to yes. With a fixed IP and host=IP address, I am pretty sure that you must turn off registration on the phone. It's useless anyway with fixed IP and just reduces reliability (as you have discovered the hard way). Asterisk periodically sends polling packets to the phone, so it will know when it is reachable and when it is not. And, the phone will still authenticate against the password, so this should not lower security at all. Stephen R. Besch ___ 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: Chaining more than one zap echo canceller?
Henry Devito wrote: A transformer would not work correctly on a phone line due to talk path being a DC voltage. 1:1 transformers only work with AC voltage. Yes it would work - otherwise hybrids, and the transformers in my Adtran interfaces, wouldn't work. The voice and ringing signals are AC voltages riding on top of a DC battery voltage and will pass the transformer quite nicely. However, that being said, don't count on a simple transformer doing the job. It would screw up other things in the interface which depend on the presence of the DC voltage (like line signaling) - and you wouldn't want a 1:1 transformer anyway. Impedance matching of this kind is non-trivial. Impedance is frequency dependent, and matching the impedance over frequency is really difficult - especially since the phone line is a distributed impedance, i.e., the R, L and C that make up the impedance are not all in one place like they are in a coil or a capacitor. Phone lines are classical transmission lines, the theory for which was worked out by Lord Kelvin in the 1890's when planning the first transatlantic telegraph cable (and was later applied, amazingly, to the transmission of nerve impulses). Line impedance refers to the characteristic impedance of the transmission line. This is the transverse impedance, that is, the impedance looking across the line. It depends upon the longitudinal resistance per unit length and the capacitance per unit length between the wire pair. Inductance is usually not much of an issue (unless LOTS of the line is coiled up somewhere or unless frequencies over 100MHz are involved). If the line is terminated in its characteristic impedance, it will appear perfectly resisistive and will optimally transmit energy into the load. If it is not terminated properly, there will be reflections down the line (i.e., echo at the far end). It is constant at a given frequency because of the distributed nature of the RLC on the line - the more distant elements make a smaller and smaller contribution. On the other hand, the longitudinal resistance, that is, the resistance measured from one end of the line to the other, is not constant but depends upon the length of the line and the resistance per unit length. Line attenuation is highly dependent on longitudinal resistance. However, none of this is really the issue with echo at the caller's end of the line. The impedance mismatch at the near end hybrid, while causing reflections and non-optimal coupling, really has its effect by introducing a phase shift in the transmitted signal. When the hybrid attempts to subtract the correct portion of the transmitted signal (the sidetone) from the received signal, echo cancellation is compromised owing to this phase shift. (As an exercise, try subtracting one sine wave from another when one of them is phase shifted. You cannot get a zero result no matter how you adjust the amplitudes.) In other words, to get your impedance matcher to work, you will need to match impedance over frequency in such a way as to eliminate any unexpected phase shift, otherwise cancellation will not be improved. In fact, if you are not very careful, you may just make things worse. Stephen R. Besch ___ 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: Configuring DIAX
Kanuri, Seshu (Company IT) wrote: I tried to setup DIAX and connect to Asterisk for the last few versions. It never actally been able to connect to Asterisk or my Other SIP Proxies like VoiceMaster. What does DIAX do actually? Is there anyone in this list who has connected to Asterisk and made call for real? Seshu Kanuri Prior to 0.9.9a, I was never able to establish voice communication. The phone registered, would dial extensions, would ring on incoming calls - just no audio stream. However, with the most recent version - same configuration exactly - now I have voice. Go figure. I agree that there appears to be no place in the help file that clearly and completely deals with configuration, other than descriptions of the fields that go something like, Enter your password in this box. An excellent example is the context field. I was under the impression that a phone didn't determine its context - this would be rather dangerous actually. I really don't know what this field does. I just set it to match the context assigned to the phone in iax.conf. Anyway, here is an abstract of my setup: iax.conf [steve1] type=friend username=steve1 secret=SomeCleverPassword auth=md5 host=dynamic context=International ;Must be a valid context mailbox=xxx ;xxx is of course your mailbox CallerID=Stevexxx ;Ditto for xxx qualify=no ;I can't make qualify work with DIAX extensions.conf ;This is in my default context exten = 1881,1,Dial(IAX2/steve1,30,Ttr) exten = 1881,2,Congestion In the DIAX registration form: Alias: An alias for your server Server: IP address of your server (see note) Username: steve1 Password: YourCleverPassword Context: International ;Must be a valid context Register: checked Name: Your Name Number: 1881 Note that I could not make DNS work for the server entry. I had to enter the actual IP or DIAX never found the server. This is OK in my case since the server has a fixed IP. For servers getting their IP dynamically, this may be a problem. Several other things that you must be careful about are the settings for the sound card. If you have more than one card, be sure that the desired one is selected. Also, be sure that the microphone (in the record controls - not the playback controls!) is not muted and is set to a suitable level. As far as I can tell, the phone delivers its audio to the sound card over the wav interface, so be sure that the volume and mute settings for wav playback are also set correctly. Stephen R. Besch ___ 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: Advice on OS Choice
Adam Goryachev wrote: On Thu, 2004-10-14 at 23:10, Alex Barnes wrote: snip I assume that I cannot actually legally install this now for NEW installs snip Oh, you can probably 'legally' install RH9 in another 20 years, I doubt there was any sort of time limit on installation for it... (not that I would know)... My understanding is that if you don't ask for or need support, RedHat (and the other distro's as well) can be downloaded and installed for free. Am I wrong about this? Isn't that the whole point of the GPL on linux? Stephen R. Besch ___ 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: Generic X100P's
The facts about Zapata hardware is that there is no buffering. If you miss an interupt, you miss data. Interupts are fired 1000 times a second or as soon as 8 bits are collected on the 8000hz phone lines. There must be at least a 1-byte buffer. There is no way you could ever get any reliability out of a card that only provided 125 us of latency overhead. That is pretty much what I said. The lack of buffering is a feature - latency is the enemy in telecomunications. That is why I said 100-200us should be acceptable. This should be redily aceivable, even with shared interrupts. 200us probably would be. And, you could probably get average interrupt latency that low. The problem is that in any real machine, even when operating in real time mode, maximum interrupt latency is much longer - sometimes in the ms range - and the standard deviation isn't that great either. If you don't believe it, hook a scope up to one of the active interrupt request lines on the bus. It's very revealing. Stephen R. Besch ___ 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: Large Scale Asterisk Migration
Stan Brinkerhoff wrote: Just as a what if... Lets say I have a 250 phone rollout. I have three incoming T1 lines (however thoes are usually setup) with say 1000 phone numbers available to me. Every phone is currently analog, but I would like to move to a VOIP based setup when the prices become comperable. What am I looking at for equipment here? How do I possibly provide 250 phone ports with the Digium 4 port pci cards? Wouldn't I need a ton of them? Thanks, Stan Unless I read your question wrong, you want to ditch the analog phones and replace them with VoIP phones. Once you get the T1 Lines into the * server(s) with standard Digium hardware (TE410's), you needn't concern yourself with analog (FXS/FXO) any more. What you need is an Ethernet tree to distribute the voice packets from the * server(s) to the phones. There's lots of recommendations in the WIKI about ethernet switches (don't use hubs) and VoIP phone hardware. I also think there are several examples of people who have built * systems with that many phones and more. If you need a channel bank to hook up some analog phones, use one of the banks you probably already have for the existing T1 lines. Just run a T1 crossover cable from the bank to a spare port in one of the Digium cards. Stephen R. Besch ___ 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: Grandstream phone price
Wolf N. Paul wrote: Except that £55 is more like $75-80 and not $35. Regards, Wolf Reminds me of a wonderful anecdote about a college english professor who, upon reading in one of his student's compositions that a character had fallen down stairs and laid prostrate on the floor, that the student had failed to make the distinction between a fallen woman and a woman who had merely fallen. Stephen R. Besch ___ 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: how to collect user entered digits
Ryan Courtnage wrote: Had not seen anything on Read anywhere else, must have been looking in all the wrong places. this is a simple solution to my problem. FYI - Read() is described on the wiki: http://voip-info.org/wiki-Asterisk+cmd+Read ___ Even better, bookmark this link: http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+commands ___ 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: budgetone 101 and buttons
Michael George wrote: I just got a Budgetone 101 and I have it hooked to my * box. I thought I'd read somewhere that we can program the buttons on these phones to send DTMF tones, thereby effectively programming them. However, according to the user's manual, they have predefined SIP functionality. My dialplan implements the festures I want (transfer, message, stuff like that), so for uniformity, I'd just like the SIP to send DTMF like my analog lines in the Digium connectors. Is this possible? As far as I know, no. Not unless you can get your hands on the Firmware sources and compile them yourself! S Besch ___ 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: dialing out and ringing issue
Info wrote: Hello: Hoping someone might know how to resolve this (probably an easy one). I have one Asterisk PBX with a single NIC and an FXO card with PSTN line attached, and one IP phone (Budge Tone 100) on the LAN. Via the phone I get no dial tone, and dialing 9, number doesn't allow me to dial out. I also need the phone to ring when the asterisk PBX is called. I have modestly tweaked the sample configs to get this far. I can check and retrieve and delete voicemail via the phone however. Any ideas? Curt The Budgetone 100 should be generating the dialtone, so this is not an * issue, per se. However, that said, the newer versions of the firmware (in the 1.0.5.xx range) suppresses the generation of dialtone if the phone does not register properly and the register option is enabled. Make sure that your phone is registering properly - see the many, some recent, posts on this subject. Note that it is possible to retrieve voice mail on an unregistered phone, but you won't be able to receive calls (because * won't know the IP address). Stephen R. Besch ___ 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: Grandstream Budgetone-102 client cannot register
Michael Cheedle wrote: Thanks for the reply Steve. Changing the conf file to 'info' looks like a good idea, but this one client still cannot register. I've pasted a debug for an authentication attempt in case it could provide more info: Are you sure that the SIP User ID and the Authenticate ID are the same and match the name you have given to the phone in sip.conf. Example: On the phone: Sip User ID:GrandstreamXYZZY AuthenticateID: GrandstreamXYZZY Password: SomeCleverPasswordorOther In sip.conf: [GrandstreamXYZZY] type=friend secret=SomeCleverPasswordorOther etc. etc. Stephen R. Besch ___ 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: Grandstream Message Waiting light
And here I was trying to figure out how to kill the blinking display :-) OK - dumb newbie award hereby rewarded to me. Thanks. And I had already checked the wiki and done what you suggested in sip.conf - so my stupidity wasn't total :-) Stupidity may be a bit strong in any case. The real stupidity was in not putting a LED under the message button in the first place. Then we could assume the intuitively obvious and wouldn't need to be confused about the multiple meaning of the flashing display. Steve Besch ___ 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: RAID affecting X100P performance...
I am. My system is a Duron 1200 with an Asus A7V266 board. I have an Athlon XP 2200 that I was planning on upgrading to. I am using the onboard IDE channels; the only cards in the server are video, ethernet, and two X100P's. No IRQ sharing. Three drives in RAID5; two of the drives are master/slave on one channel, the other shares a channel with a CD drive. I'll run the same tests you did and report back with the results. Unless I read the manual correctly, it is not possible to have the configuration you specify without shared interrupts on this motherboard (Asus A7V266E): Slot1: uses INTD, which is shared with the RAID, USB and slot 5 Slot2: uses INTA, Shared with AGP, onboard audio and onboard LAN Slot3: Uses INTB, Shared with AGP Slot4: Uses INTC, The only unshared slot Slot5: Uses INTD, Shared with slot 1, RAID and USB Looking at the manuals for several other MOBO's, similar results situation seem to exist. My guess is that Slot1 contains one of your X100P's, so when the RAID is enabled, interrupt response time goes into the toilet. The point is, as long as motherboard manufacturers are going to take the easy (lazy?) way out, and share slot interrupts with each other and with other hardware, you are pretty much stuck with not putting your RAID on the same box as the X100P. Something which seems never to be considered about echo is the nature of the true echo path. It obviously includes the delay over the analog wire, if there is one, plus the sum of all the packet delays on the network. What is forgotten is that it also includes the delay through the CPU, which, in turn includes the delay through the cancellation process. If the delay is always constant, cancellation becomes much more straightforward. For any real world echo canceller to work, it must be able to adapt to variable delays. The problem is, that if the delay adaptation response is too fast, cancellation becomes unstable, if it is too slow, cancellation is not effective. Now, look at the problem the T100P suffers as part of the echo cancellation stream. Each time it needs service, the time to be serviced changes based on random fluctuations in interrupt service time, even if the interrupt is given highest priority (i.e., a low interrupt number). Typically, this fluctuation is not too bad, at least in terms of the requirements of echo cancellation. Now, put a shared interrupt on the T100P. Even worse, you assign it to share with a disk controller, which by definition should be given a low priority (that's why the IDE interrupts are always 14 and 15!) You have now just put horrific delay fluctuation in the cancellers path. Everything gets priority, including the disk if you are unlucky. These wild, and large, fluctuations in delay will kill any echo canceller, no matter how good it is - the exception of course being an independed canceller that processes the echo before it even gets to the X100P and the asterisk box. Similar logic applies to the network card, since it also lies in the cancellation stream and it's interrupt response time can significantly affect path delay. The solution? Well, I'm guessing here, but you might be able to get a big improvement by using the one unshared interrupt for the X100P, use a PCI video card to free up two of the shared slots and put the second X100P in slot 3. Finally, put your lan (if it's not onboard) into slot2. Finally, and this may be the most important thing, don't let the motherboard automatically assign interrupts to IntA-IntD. Go into the BIOS and force it to assign the highest priorities available to the T100P and the net card. Also, disable any unused hardware you're not using which may have an interrupt assigned (eg, the USB controller). For the ASUS at least, the priorities assigned to the interrupts are listed in a table in the user manual (p32). Stephen R. Besch ___ 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: RAID affecting X100P performance...
Scott Laird wrote: In the context of Asterisk, where disk I/O is either logging or voicemail, buying a 3ware card and a pair of IDE drives seems like a decent business decision. I agree. Even more to the point, when using T100P's which need high quality, fast service from the CPU and the host bus, I would recommend putting your RAID in a separate box. Any cheap MOBO with a raid card, software raid, onboard raid controller, etc., should suffice since the only thing it will be doing is RAID and the maximum I/O rate will be 10MB per second or so, limited by the ethernet speed. If you use internet for a voice channel, make sure that you use a separate NIC to talk to the RAID, and assign it a lower interrupt priority than those used by the telephony hardware. Stephen R. Besch ___ 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: PRI dead in USA?
Andrew Kohlsmith wrote: On Tuesday 20 July 2004 18:18, George Pajari wrote: In spite of what my learned colleague implies above, there is more to Canada than Ontario (Bell's territory). Please retract your statement that I implied anything of the sort; I never even mentioned the province I was in, nor do I harbour any kind of cold hostility toward the western provinces as you seem to imply here. We also have Telus and ATT and Sprint and MCI... hell even Group Telecom, but unfortunately in my little town you can't get a PRI from anyone but Bell; the others wouldn't even return my calls. Out here in the West (Vancouver -- rarely acknowledged to exist by Torontonians) PRI B channels are a lot more expensive than POTS. I'm not from Toronto, nor any other major city for that matter. Honestly though, was this kind of attack on me (or other Ontarians) necessary? Could you have not just stated your situation and pricing from your point of view without taking a shot at me or where I live? -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Some people just have bristly whiskers! Steve Besch ___ 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: Updated Grandstream configurator
Maron Kristófersson wrote: I was even considering going further and writing a crossplatform or a webapp for configuring. However I was thinking if someone has written some notes on the config file specification See: http://www.mail-archive.com/[EMAIL PROTECTED]/msg43052.html Also, refer to the sources of GSConfigure. that could save a lot of time. I have no intention of competing with gsconfigure since I think it's an excellent What? Competition is good! Stephen R. Besch ___ 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: Updated Grandstream configurator
I now have a batch mode installer for GSConfigure which works on at least 2 of the machines here. If you are having trouble getting the VB installer version to work, give this a try. It's at http://www.acsu.buffalo.edu/~sbesch Stephen R. Besch ___ 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: VoicePulse changes
Remco Barende wrote: On Thu, 15 Jul 2004, Chris Glover wrote: On Thu, 15 Jul 2004 [EMAIL PROTECTED] wrote: Note The previous method for terminating IAX2 calls using Connect! will cease to be available at midnight (GMT) on August 15th, 2004. The message I got was at 1:51 AM EST. That means I was given negative 5 hours and 51 minutes to make this change. How did you get -5 hours? I make it 31 days! :-) Different time zone ?? :) Maybe if you circle the globe enough times, crossing the international date line each time, of course, it would be possible to get to August 15th yesterday ;-) SRB ___ 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: Updated Grandstream configurator
Maron Kristófersson wrote: I'm very close to making this work in the crossover wine emulator on linux. Currently I am getting an error when trying to download the config directly from an ip address. See attached snapshot for details. Post me the snapshot directly. It won't come through the news readed I am using for *. When installing the program I had to choose win2k as the emulated OS. That's not surprising. Several people have been unable to get it working on WinXP. I'm going to try and build it on XP and see what happens to 2K compatibility. Regards, Maron Kristófersson Stephen R. Besch wrote: Greg Boehnlein wrote: On Thu, 8 Jul 2004, Neil Cherry wrote: Stephen R. Besch wrote: The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I could find inn an effort to solve the VB setup bug, so, hopefully it will no longer send anyone through multiple restarts. You should have at least SP3, or even better, SP4 on Win2k. I believe it will run on Win9x, but I have not tested it and can make no guarantees. Thanks, I've been having real trouble with those stupid DLLs. I can't upgrade some of them no matter what I do (WIN2K)! I downloaded and installed it on a Windows 2000 server box. The install complained about me having a newer .dll file than what was being installed, so I chose to keep my version. Seems to work fine. Does this support the new 1.0.5.x firmware train? I have tested against 1.0.5.0 with success. I know that there has been at least one new option added in some of the later revisions. As soon as I am convinced that they are stable, I'll load one up and test it. Adding another parameter/option pair is easy. Stephen R. Besch ___ 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] Re: Updated Grandstream configurator
Maron Kristófersson wrote: OOPS! Ignore that last post, I found the snapshot in your next post. SRB ___ 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: Updated Grandstream configurator
Steve wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 08 July 2004 05:45 pm, Stephen R. Besch wrote: The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I could find inn an effort to solve the VB setup bug, so, hopefully it will no longer send anyone through multiple restarts. You should have at least SP3, or even better, SP4 on Win2k. I believe it will run on Win9x, but I have not tested it and can make no guarantees. Steve Besch The bad part is that starting with SP2 on w2k ms EULA has changed to include your agreement to let microsoft not only see, what you have on your computer, but also install software on it. This has caused a big corporate hold on updating beyond SP2. The medical industry in particular is having a hard time, as ms has not signed a non disclosure to have access to personal medical information. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin Since I am quite sure that the program will run without updating any of the dll's, what I should do is simply register them with regsvr32 from a batch job and bag the VB6 installer altogether. Before I do that though, can anyone tell me if regsvr32 ships with standard Win2k/WinXP? Stephen R. Besch ___ 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: Updated Grandstream configurator
Maron Kristófersson wrote: and the attachment is here :) Maron Kristófersson wrote: I'm very close to making this work in the crossover wine emulator on linux. Currently I am getting an error when trying to download the config directly from an ip address. See attached snapshot for details. Maron, Sorry about the delay getting back to you about the object creation error indicated in the snapshot. I am almost certain that the error is deriving from the creation of http objects by winhttp.dll. Microsoft's implementation of HTTP stuff for VB is really lame. I couldn't make the standard HTTP control work at all with the Grandstreams, and winhttp would only work bug free with post operations. Get operations were transmitted, but crashed the program if I tried to access any of the data that was supposedly returned. It doesn't surprise me that emulation of the winhttp's object creation also fails. This also means that the phone reboot code will fail, since I also create an http object for each booting phone. One thing you should check is to make sure that winhttp.dll is in fact supported by wine and that you have the support installed. I wish I had better suggestions. Is there any way to splice a different http handler onto the program using wine? One possibility is to scrap the winhttp com object altogether and write the http code using the c/c++ interface from VB, which doesn't use object creation. This would make programming the http stuff a lot harder, but might avoid the problem. Then again, if winhttp is not supported by wine, this won't help. I'll keep thinking about it, but for now, I'm at a loss. Steve Besch ___ 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: Updated Grandstream configurator
Greg Boehnlein wrote: On Thu, 8 Jul 2004, Neil Cherry wrote: Stephen R. Besch wrote: The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I could find inn an effort to solve the VB setup bug, so, hopefully it will no longer send anyone through multiple restarts. You should have at least SP3, or even better, SP4 on Win2k. I believe it will run on Win9x, but I have not tested it and can make no guarantees. Thanks, I've been having real trouble with those stupid DLLs. I can't upgrade some of them no matter what I do (WIN2K)! I downloaded and installed it on a Windows 2000 server box. The install complained about me having a newer .dll file than what was being installed, so I chose to keep my version. Seems to work fine. Does this support the new 1.0.5.x firmware train? I have tested against 1.0.5.0 with success. I know that there has been at least one new option added in some of the later revisions. As soon as I am convinced that they are stable, I'll load one up and test it. Adding another parameter/option pair is easy. Stephen R. Besch ___ 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: sample config file for GS BT101?
Stephen J. Wilcox wrote: I was wondering about that too.. Following the instructions on that page for config did not work for me. Setting up a config file like the sample one made no difference to the phone (I can confirm it did tftp it okay). Also the method references md5 checks and I dont see that at all. I tried the downloads, we wouldnt do this from windows so need to know how to do this to write for *nix but I couldnt get the windows app to run on XP/2000 machines altho apparently it will run on 98 but I wasnt able to test that with a phone. So - is it literally just supposed to be a case of creating a blah=blah style config file at mac.txt ?? I note not all the options are listed in the sample, what about the others? And finally.. why doesnt this info appear to be available from the manufacturer, surely we shouldnt be reverse engineering? Steve Here's the format of the file. It works perfectly in the GSConfiguration program I have posted to the list. I believe that this format is complete. Each entry is Function, length, format. File Length, Dword, Bigendian The length of the file in WORDS. Length is null padded to even number of words Checksum, Word, BigEndian Value added which makes the 16-bit sum of the file equal to 0 MAC, Byte[6] Unique hardware address of SIP device. CRLF,CRLF 4 bytes set to 0D,0A,0D,0A Body 8-bit ASCII string which contains the phones parameter data in am delimited list. Each parameter is of the form Px=value, where x is an integer and value is the contents of the parameter. The parameter may be null or missing. If the parameter is null, the coresponding value is set to null. If it is missing, the currently set value in the phone is used (that is, only those parameters present in the file are changed). The order of the parameters appears to have no significance (except for gnkey, maybe - see below). Example: ...P12=110P17=0P18=0P19=0P20=0 There is no separator either before the first parameter or following the last. In addition, some preliminary testing suggests that parameters which have no corresponding phone parameter are ignored. This theoretically makes it possible to include user customized data in the cfg file to be used for other purposes, such as storing a person's room number, etc. This could be quite useful in cfg management programs. Example: ...P20=0Room=Oval Office... gnkey=0B82 This must be specified as the last parameter. As with the others, it is specified as text, and separated from the previous parameter with the separator. I have not tested this, but I assume that it must appear last in the list. Terminal Null This is present only if the parameter text string ends on an odd byte boundary. It is added to make the file an even number of words long for the checksum routine. I suspect that it really can be any value you like (I have not tried this, by the way), as long as it is included in the file length calculation so that it will not turn up as a random value, or, worse, cause a file read overflow when the phone attempts to process 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
[Asterisk-Users] Updated Grandstream configurator
The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I could find inn an effort to solve the VB setup bug, so, hopefully it will no longer send anyone through multiple restarts. You should have at least SP3, or even better, SP4 on Win2k. I believe it will run on Win9x, but I have not tested it and can make no guarantees. Steve Besch ___ 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: Grandstream CFG file generator
Tomas Prybil wrote: Stephen R. Besch wrote: snip This is the install package for the program. Running setup will install the program into Program Files\SachsLab\GSConfigure and put a shortcut in the start menu under Phones. The sources are installed to the application directory in a folder named Source. If you have never installed a VB6 setup package, then wyou will very likely get the following message: Setup cannot continue because some system files are out of date on your system. Click OK if you would like setup to update these files for you now. You will need to restart Windows before you can run setup again. Click cancel to exit setup without updating these system files. You can safely click OK to update these files. Also, during the install process, if any of the files already on your system are newer than those in the package (you will be notified), you should opt to keep the ones already on your system. If by chance you happen to already have Bruce McKinney's Windows type library on your machine, then you should install the one that comes with this package, since it includes some constant definitions that are not in the original. If, in the extremely unlikely event that you have your owm custiomized version of the type library, then you should contact me off list if you want to play with the sources. However, the files doesn't gets updated or something else. The install process allways ends with that out of date message. Looking at you SETUP.LST the following files are missing from my PC: [Bootstrap Files] [EMAIL PROTECTED],$(WinSysPathSysFile),,,7/15/00 12:00:00 AM,101888,6.0.84.50 File2=OK [EMAIL PROTECTED],$(WinSysPathSysFile),$(TLBRegister),,6/3/99 12:00:00 AM,17920,2.40.4275.1 File4=OK [EMAIL PROTECTED],$(WinSysPathSysFile),$(DLLSelfRegister),,3/8/99 12:00:00 AM,164112,5.0.4275.1 [EMAIL PROTECTED],$(WinSysPathSysFile),$(DLLSelfRegister),,4/12/00 12:00:00 AM,598288,2.40.4275.1 File7=OK BR /t After some research, I have discovered that this is a bug in the VB setup system. I have been able to duplicate the problem on machines that are not up to date (i.e., don't have SP4 installed). For some reason, several files are not updated at all ( in partucular, msvcrt.dll and scrrun.dll). Since this program does not need the new versions, their entries can be safely removed from the list of installed files. If anyone else has this problem, contact me off-list for details. Also, there is a new version of the program that fixes several bugs. See the post elsewhere on the list. Stephen R. Besch ___ 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] Updated version of Grandstream cfg file generator
Several bugs were reported in the first release version, which are now fixed: 1) The file generator was losing the filename and not successfully generating files when using the MAC address. 2) The path initialization code was not working correctly. Also, I have appended the version number to the main window's title and added an option to show phone passwords in plain-text for those people who don't have someone looking over their shoulder all the time. Finally, the booting code has been changed to take the phone password directly from the configuration file, eliminating a lot of typing if the phones don't all have the same password. The only time a password is requested is when it can't be extracted from the configuration file for some reason. The new version (1.0.25) should be available shortly at http://asterisk.4gurus.org/ Stephen R. Besch ___ 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: Really basic stuff :(
Gavin Hamill wrote: Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ Host' so all incoming IP traffic (even AH/ESP for IPSec etc.) goes directly to that machine. I am not doing any firewalling, nor is my ISP. I've made my configuration as superficial as I can to ease diagnosis: [EMAIL PROTECTED]:/etc/asterisk# ls -l -rw-r--r--1 root root 104 Jun 23 21:21 extensions.conf -rw-r--r--1 root root 164 Jun 23 19:25 iax.conf -rw-r--r--1 root root0 Jun 22 15:36 modem.conf -rw-r--r--1 root root 387 Jun 23 21:22 modules.conf -rw-r--r--1 root root 363 Jun 23 21:19 sip.conf -rw-r--r--1 root root0 Jun 22 15:36 voicemail.conf I know that this is not related to your ultimate question, but I would not recommend giving read access to everyone. Even if you have guest disabled, this still leaves you vulnerable to snoops discovering your configuration. With that in hand, they can make phone calls on your dime. I would change the access rights on all of these files to 640. Stephen R. Besch ___ 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: Grandstream CFG file generator
Stephen R. Besch wrote: I've just finished a general purpose configuration utility for the GS phones: 1) Generates files from scratch (using MAC), from HTML config listing, or by directly downloading from the phone. 2) Does multiple simulteneous edits. 3) Can reboot as many or as few phones at a time as you like. The zipped install package can be downloaded from: http://asterisk.4gurus.org/ I have updated the readme to clarify some issues with the install process, but the updated text will not be in the zip file just yet. For this reason, I have included the body of the readme below Stephen R. Besch == readme.txt: This is the install package for the program. Running setup will install the program into Program Files\SachsLab\GSConfigure and put a shortcut in the start menu under Phones. The sources are installed to the application directory in a folder named Source. If you have never installed a VB6 setup package, then wyou will very likely get the following message: Setup cannot continue because some system files are out of date on your system. Click OK if you would like setup to update these files for you now. You will need to restart Windows before you can run setup again. Click cancel to exit setup without updating these system files. You can safely click OK to update these files. Also, during the install process, if any of the files already on your system are newer than those in the package (you will be notified), you should opt to keep the ones already on your system. If by chance you happen to already have Bruce McKinney's Windows type library on your machine, then you should install the one that comes with this package, since it includes some constant definitions that are not in the original. If, in the extremely unlikely event that you have your owm custiomized version of the type library, then you should contact me off list if you want to play with the sources. Stephen R. Besch [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstreams randomly go busy with Asterisk?
Kyle Hagan wrote: Brian Buhrow wrote: Hello. I've seen this behavior. What happens is that the Grandstreams forget to continue registering with Asterisk after a while. I bet when you find this happening, that sip show peers doesn't show ext/ext ip address for the one that isn't working. You can work around the problem by explicitly telling Asterisk how to dial the GS by giving it an explicit IP address in its sip.conf extension entry. Alternatively, you can upgrade the Grandstream to a newer load of firmware. I'm running 1.0.4.68 on my HT286, and it seems to behave much better. I got my firmware load from: http://www.voiptalk.org/products/download/ They seem to have 1.0.4.63, and 1.0.5.0, but not 1.0.4.68 anymore. Hope that helps. -Brian I have version 1.0.5.0 in my GS and still does the same thing. But have not setup the IP address in SIP.conf I would like them to be dynamic. What a scream! Hasn't GS removed the Don't Register option on the 1.0.5.0 software? If it still has amnesia about registering, this is a real coup. ___ 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] Grandstream CFG file generator
I've just finished a general purpose configuration utility for the GS phones: 1) Generates files from scratch (using MAC), from HTML config listing, or by directly downloading from the phone. 2) Does multiple simulteneous edits. 3) Can reboot as many or as few phones at a time as you like. I would like to offer it to the list, but there are 2 issues: 1) I want to GPL it first, if possible. Problem is I am not sure exactly how to go about this. 2) I don't have any means to distribute the files (other than e-mail, which I really don't want to to. So, if someone could brief me on the GPL issue, and (perhaps someone else) offer a distribution point, it's free for the asking, VB sources and all. Stephen R. Besch ___ 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] Grandstream Ringtones on a per phone basis
I have just successfully got the TFTP file remapping to work such that I can have unique ringtone files for each and every extension. I added the following to my server_args line in the xinetd configuration for TFTP: -m /home/asterisk/grandstream/ringmap.cfg Now the entire line reads: server_args = -v -s /home/asterisk/grandstream -u asterisk -m /home/asterisk/grandstream/ringmap.cfg (There is no line break in the actual file!). When debugging, you can use more v's in the verbosity option. Then, in the TFTP root, I add the file ringmap.cfg, with access rights set to 660 (so that the TFTP server won't serve the file up). That file contains the following line: ri ring(.)\.bin ring\1_\i This tells TFTP to replace any file requests that look like ringx.bin with ringx_yyy.yyy.yyy.yyy, where yyy.yyy.yyy.yyy is the phone's IP address. So, for example, if 192.168.10.100 requests ring1.bin, TFTP will get ring1_192.168.10.100 and serve it up as ring1.bin, effectively lying to the phone. Then you only have to generate your ring files and place them in the TFTP root folder with access rights 664. Works like a charm! Stephen R. Besch ___ 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: Grandstream Early Dial
Aaron Martin wrote: Has anyone managed to get Early-Dial working with the grandstream phones? On my older phones running firmware 1.0.3.X it works fine, but it doesnt work on the newer versions.. Don't waste time trying. I'm even surprised that you could get it to work with 1.0.3.x. In my experience, if the dial string was longer than 4 or 5 digits, GS pooped out. It's even worse in the later revisions, pooping out after 3 digits. It still is not fixed in 1.0.5.0. You could use it if all of your dial strings are 3 0r fewer digits - not much use really. GS knows about the problem, has verified on my * server, has indicated that they will fix it. It just requires patience on our part. The fix is apparently not a very high priority - true really, since it's failure is merely a convenience issue of not having to wait for the last key timeout. Stephen R. Besch ___ 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: GS HandyTone Issue
Stephen Rosebush wrote: I just got myself a GS HandyTone and it works great, it was a breeze to setup. My only issue is I seem to be hearing a humming noise on the line when I am in calls.. You have a short (or leak) to ground somewhere in the analog line. S Besch ___ 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: Grandstream 1.0.5.0 Firmware: SIP Register option gone
Brian Capouch wrote: FYI to all you Grandstream users out there. I just fetched and installed the 1.0.5.0 firmware, and it appears they have removed the option to either do or not do SIP registration. Now it appears that one is going to register with the server specified in the SIP Server field, without any ability to disable it. Bug or feature? Just thought I'd put this out there and see if anyone knows anything about it. I doubt it's a bug in the phone because you cant disable the registration with either the config file, or by loading the 1.0.5.0 with the 1.0.0.31 HTML page - meaning that the change has been made in multiple places, not at all characteristic of a bug. The bug is in the brain cells of the engineers at GS, who, by all rights, have broken the most cardinal rule in computer science: Never, Never, Never alter an interface in such a way as to break a significant number (more than 1?) of installations. Now, mind you, this is Beta code, and the option may be reinstated before the code goes to release. Nevertheless, the cavalier way in which they have broken many installs with this removal, while leaving such rarely used things as off-hook auto dial, the almost useless dial plan field, the broken early dial option and crippled ring tones suggests that someone in the organization really needs a talking to. I know that GS monitors this list. I just hope that they take this seriously. Breaking the no register option is a really serious, and idiotic, mistake. They really are obliged to fix this as soon as absolutely possible. Stephen R. Besch ___ 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: Grandstream 1.0.5.0 Firmware: SIP Register option gone
Brian Capouch wrote: Tomas Prybil wrote: Brian Capouch wrote: FYI to all you Grandstream users out there. I just fetched and installed the 1.0.5.0 firmware, and it appears they have removed the option to either do or not do SIP registration. Now it appears that one is going to register with the server specified in the SIP Server field, without any ability to disable it. Any other features you've empirical found out but that? Not as far as I can see, and the configuration files from the older beta (4.68) still work (other than simply ignoring the do not register option). I had to quick revert back, because I use static IP assignments on my phones and the 1.0.5.0 software quickly caused a fair amount of havoc on my system. It does revert fine, btw, by just putting the older stuff on the tftp server. So I'll probably rev up later on and play around some more. It still didn't load the ringtones, and I have to wonder if the speculation I've read on the list that some phones won't load them might not be correct. I note the tones with 1.0.5.0 are all files 64Kb. They claim to have altered the firmware to accept larger files. Nevertheless, there really doesn't seem to be any good reason to go to the 1.0.5.0 version anyway. It hasn't fixed any of the outstanding issues (at least those related to use with *, or added any really useful functionality. Stephen R. Besch ___ 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: Transfer with Budgetone
Tony Hoyle wrote: Stephen R. Besch wrote: Not as far as I know, at least not exactly the way you have outlined it. Try this: 1. call comes to you 2. You hold the call and call other person. 3. You say Someone wants to talk to you, OK, thanks 3a. Other person then hangs up. 3b. You flash back to the original caller 3c. You tell them that you are transferring the call 3d. You transfer the call using the transfer feature on the phone 4. You hangup and first person is transferred to other person? Ugh. So Asterisk doesn't handle transfer? Every company phone system I've ever used has not required 3a-3d. It looks like a real hack to do so. It anyone working on implementing this? Tony Actually, I think this problem is with the Budgetone, not *. While I haven't tried it myself, from what I have read in the list, the * transfer functionality works just fine. Stephen R. Besch ___ 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: Grandstream ringtone maker (was Re: Grandstream v1.0.4.68 firmware)
Stig Hess wrote: Now if I could only get my GS phones to load the ring tone files. The TFTP log shows all the requests for the usual boot files and the cfg files but NO requests for the ring tones, not even file not found responses. I can't believe that this is the tftp server. I have tried it on at least three different phones, purchased in 2 different lots and still no luck. Maybe the phones just don't like me. I have exactly the same problem. Could there be different hardware versions? Stig Can't say I haven't wondered that myself. Given that there are no serial numbers visible on the phones, I suspect that one could use the MAC address as a serial number, since these are probably allocated sequentially. Only the last 6 digits should change. My first phone (bought as an evaluator) has last 6 digits of 002175. The second shipment has numbers in the range of 002935 and up. It would be interesting to know if the MAC's from the phones that work are significantly higher than these. Stephen R. Besch ___ 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: Grandstream ringtone maker (was Re: Grandstream v1.0.4.68 firmware)
Tony Mountifield wrote: Make sure the ring tone files are no bigger than 65536 bytes. Earlier versions of my program didn't check for this, but the latest one does. That's potentially important information, but even still, the tftp log would show the phone requesting the file, even if it rejects it later for being overly long. Stephen R. Besch ___ 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: Adtran TSU 600
Bartosz Jozwiak wrote: Hello, Did anybody successfully tried upgrade Adtran TSU 600 to firmware which is working properly with T100P and asterisk ? B. Yes, but it was a while ago (last August). I currently have the TSU600 with 2 FXO/1 FXS cards running on a T100P with the only problem being that the FXS card is a little flakey, but this has no bearing on the T100P. I just downloaded the firmware and the stripped down version of T-Flash that Adtran provides for flashing the firmware. Once the serial link to the TSU was up and running properly, the firmware update went without a hitch. Stephen R. Besch ___ 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: Transfer with Budgetone
Sergio Serrano wrote: Hi all, I try to do next transfer: A person contact with me, I would like transfer to other person in next manner. I call to other person and when I say who wants talk with him I hangup phones an call is redirect automatically to other person: 1. call to me 2. Hold the call and call to other person. 3. I say Anyone want talk to you, OK, thanks, 4. I hangup and first person is directly redirect to second person? It is possible with asterisk and budgetone phones? Sergio, Not as far as I know, at least not exactly the way you have outlined it. Try this: 1. call comes to you 2. You hold the call and call other person. 3. You say Someone wants to talk to you, OK, thanks 3a. Other person then hangs up. 3b. You flash back to the original caller 3c. You tell them that you are transferring the call 3d. You transfer the call using the transfer feature on the phone 4. You hangup and first person is transferred to other person? Stephen R. Besch ___ 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: Grandstream ringtone maker (was Re: Grandstream v1.0.4.68 firmware)
Philipp von Klitzing wrote: Hi! Excellent!! Philipp See near the bottom for the interesting bit :-) OK, while composing this post I decided to write a perl program to read a uLaw stream on standard input and create a suitable header, writing the result to an output file. It can be found at http://www.softins.co.uk/makering.pl.txt Now if I could only get my GS phones to load the ring tone files. The TFTP log shows all the requests for the usual boot files and the cfg files but NO requests for the ring tones, not even file not found responses. I can't believe that this is the tftp server. I have tried it on at least three different phones, purchased in 2 different lots and still no luck. Maybe the phones just don't like me. Stephen R. Besch ___ 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: Grandstream tftp cfg.txt format
Maron Kristófersson wrote: Hello! I've been reading through the archives on this list for the last 8-10 months. There are some reports on success with tftp autoconfiguration with a given cfg.txt format but really vague. Has anybody successfully done this without using GAPS, or has anybody got a correctly formatted cfg.txt file that works (from GAPS). I would be happy to write a script or a java program that creates such a file, but I need the format to do that. Regards, Maron Kristofersson I suspect that the txt version of the cfg file is used as input to GAPS, not input to the phone. What we really need is a handful of working cfg files that have already been compiled by GAPS into the loadable binary format that the phone probably wants. If each file had only one item changed, and were accompanied by a detailed description of the corresponding phone setup, then the file format could be decoded, of course, providing that the file format is not encrypted (I doubt this, however). The problem will be getting the required files. GS asked me to sign an NDA before they would even consider letting me experiment with their GAPS system. I suspect that everyone else has had to do the same. Stephen R. Besch ___ 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: Grandstream v1.0.4.68 firmware
Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? Stephen R. Besch P.S. Grandstream, if you are listening, then Early Dial is still broken! It's been many months now to fix what apparently is just a counter bug. Come on, let's get this fixed. ___ 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: Grandstream v1.0.4.68 firmware
Jeremy McNamara wrote: Stephen R. Besch wrote: Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? Didn't you hear you've gota purchase their $100,000,000 provisioning tool to enable ringtones. Jeremy McNamara Yes, well we can at least be thankful that they haven't adopted the Cisco model. Then we would need to pay them a licensing fee each time the phone rings. SRB ___ 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: Grandstream v1.0.4.68 firmware
Thomas Gallaway wrote: Jeremy McNamara wrote: Stephen R. Besch wrote: Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? Didn't you hear you've gota purchase their $100,000,000 provisioning tool to enable ringtones. My ringtones just work on all the grandstream's :-) Bizarre, really bizarre. SRB ___ 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: Grandstream v1.0.4.68 firmware
Thomas Galloway wrote: Stephen R. Besch wrote: Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? I just slapped them all onto my TFTP server and they all load fine. Then on the bottom I can choose between the 3 ringtones and tell it to ring a certain ringtone if coming from a certain caller id. I just do not like the ringtones (piano). It'd be great if there was a way to upload own ringtones but I can not seem to be able to find out how to edit the files. That has not been my experience. I do indeed get the option to associate each ring tone with a given caller ID (which in my estimation is a really stupid implementation anyway - the real value would be in associating each line on the 2 line GS with a different ring tone. The caller ID already tells you who is calling). However, I can put anything I want into the text boxes and nothing happens - I always get the system ring tone. And, what are those stupid little radio boxes for. No matter which one I check, when the screen refreshes it defaults back to the System Ring Tone. Here's what they look like (the o's are supposed to be radio boxes): o System Ring Tone o Custom Ring tone 1, used if incoming caller ID is (Text Box) o Custom Ring tone 2, etc. What's this supposed to mean? It implies that if I select one of the custom ring tones, then the phone will ring on the matching CID, otherwise, it won't ring at all! This feature really needs work. I hope it doesn't wind up like the useless Daylight Savings Time option, which you may have noticed does not pay any attention to the date so you have to log into each and every phone and change the option anyway (please, correct me if this has been fixed). Why bother? I can just as easily change the time zone and get the same effect. GS is obviously targeting their phones for the consumer market where a customer has 1, maybe 2 phones and this kind of thing is irrelevant. The whole concept of their overpriced provisioning system is rather a funny joke in this context, and a rather pretentious one at that. SRB ___ 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: Grandstream v1.0.4.68 firmware
Thomas Gallaway wrote: Brian Capouch wrote: Thomas Gallaway wrote: My ringtones just work on all the grandstream's :-) Do the URLS for the ringtones at the top show up as something other than all zeroes? I've fiddled with this until blue in the face, and the ring sounds just like the ring it had before. This is with 1.0.4.68, but it was no different with the earlier supposedly ringtone enabled version. *Product Model: * BT100 *Software Version: * Program--1.0.4.68Bootloader--1.0.0.16 HTML--1.0.0.31VOC--1.0.0.5 *Custom Ring Tone: * ring1--1.0.0.0 ring2--1.0.0.0 ring3--1.0.0.0 (all zeroes means unavailable or unsupported) -- Thomas Well in my case, the ring versions are all 0.0.0.0 no matter what I do. Could you also post the exact spelling of the binarys on the tftp, including capitalization, access rights and access mode. Mine are ring1.bin, ring2.bin, ring3.bin, all lower case, owned by root, in the asterisk group and with r/w mode=rw_rw_r__ (664), which is the same as all the other items on the server, which do in fact get loaded. Also, I've checked the binaries and they do in fact have version 1.0.0.0 embedded in the file (look at hex offset 6, which is where the version signature of all the GS bianries is located). I also can connect to the tftp server from another machine and successfully get ring1.bin. Perhaps the binaries that came with my copy of the firmware are corrupted. Maybe you could zip up the tones you are using and post them to me so I could see if these fix the problem. SRB ___ 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: Grandstream v1.0.4.68 firmware
Serge Oleinikov wrote: I was trying to replace the header. But looks like header contains some kind of CRC The format the rings are at are after what I found out uLaw compressed 8bit 8000hz mono samples. But they also have a header infront of the file. I will play arround with it later. Maybe there is a way to chop off the header of the ones that come with it and put it infront of a regular file. I did a little detective work and here is a summary of what I found out. In version .63 of the firmware, the ringtone files are all nearly identical (at least they contain the same audio data stream). From this, and from the format of the hex data in the header, I was able to discover the location and format of the checksum. The idea is that the only differences in the files were the file name field, the checksum word and one other value (the 00C8 at offset 26). From this it was possible to compute the difference in checksum expected and show that the only value in the file that changed appropriately was at offset 2. It also revealed the means of calculating the checksum. Here is a partially decoded header: Hex Offset Typical Value Function 00 ? Always zero (6 sample files) 02 7F90 File length in 16-bit words (bigendian) 04 3450 Checksum (see below) 06 0100 Version number 0A 07D4 Always this value (6 samples) 0C 0419 or 0505 ? 0E 82A, 140B, 142C ? 10 Text filename (eg ring1.bin) 19-25 0's? 26 0 or 00C8 ? 28-FF 0 ? 100 0100 **See below 102 7F90 repeat of length 104-127 0's? 128 Text String describing file 147-1FF 0's? 200-end Audio Data The checksum is the value that must be put into location 2 so that a 16-bit sum of the entire file, ignoring overflow, is exactly 0. It is essentially the negative of the sum of the file computed with a zero value in the checksum. The value in location 100 is interesting. I suspect that it is either the length of the buffer which follows and contains the file comment, or it is simply the offset to the data stream. Since is sits at location 100, it is impossible to tell which. Some further experimentation might resolve the issue. If anyone can fill in the remaining fields, it would be cool. Hope this helps. Stephen R. Besch ___ 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: Transfering with Grandstream Phones
MPlus wrote: I have the same problem with 2 ATA-286s, DTMFMODE=info and Dial command with Tt options. Only the caller is able to transfer the call with the # key. The callee is not able to transfer the call using # key, unless the codec is ULAW and the DTMFMODE is inband. I suspect the problem is due the GS unit because I failed to detect any DTMF INFO packets going into asterisk from the callee using ethereal. DTMF INFO packets were detected from the caller, though. MPlus - Original Message - From: Paul Tyreman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 09, 2004 2:09 AM Subject: [Asterisk-Users] Transfering with Grandstream Phones Hi, I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have tried every conbination of T and t in the extensions.conf file, but all to no availe ! Can anyone help ? Thanks, Paul. ___ 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 Make sure that you don't have either the Use # as dial key or Send Flash Event options set to yes. They must both be set to NO for transferring to work on the GS Stephen R. Besch ___ 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: -- MARK --
You guys are funny! Mark Spenser! Haha! I knew immediately it was from Mark Musone! Wasn't Mark Spenser a medieval poet or something? ___ 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: Opinion poll: best SIP phones for asterisk?
Iain Stevenson wrote: I'm running 4.50 because of adverse reports of 4.53 etc. Is abbreviated dialling (aka Early Dial) working yet - it's been out of commission for most firmware from 35 - 50 releases. No. Tested it last week on '4.54 - still broken - but somewhere along the line they fixed the call waiting ring tone volume. Has anyone else noticed a strange problem with call waiting in 4.54 - the sound becomes one way (remote end can't hear you but you can hear them) about 5-10 seconds after switching to a call coming in on the second line. Whatever the problem, hitting flash again restores two way sound. Steve Besch ___ 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: Asterisk SIP + Grandstream 100 + sip.conf phone HELP
Sorry about the post to the wrong level of the thread, but something was wrong with the first copy of the message (i.e., my mail reader wouldn't display it). Comments are inline. I tried Stephen advice and it did not work. I stil got the 404 error [general] dtmfmode=rfc2833 This does not match the selection used in your phone, and ironically, is the only choice that does not seem to work on the GS phones. Use inband or info and make sure that you set the phone the same way. ;[snomsip] ;type=friend ;secret=blah ;[pingtel] ;[cisco] ;[cisco1] You might consider deleting all of these unused bits from your file, or at least from the email before you send it. If you need them later, you can always copy and paste them back from a reference copy of the file. [1001] type = friend context = default secret = gol host = dynamic Unless you have a good reason for using the dynamic option, I would not use it. In your case, the phone's IP is Hardwired, and private to boot. Just put the IP in after the host=. You also avoid the (possibly still present) grandstream bug which loses registrations from time to time. callerid = STREAM-1001 1001 ;dtfmmode=inband Ironically, this is what you used on the phons. Why is it commented here? canreinvite=no defaultip=192.168.0.105 [1002] Same for phone 2 This is the configuration of my SIP-phones: outboundproxy=null outboundproxy_port=null If all else fails, put your server IP in here! Use default port registration_expiration=10 You may find registration to be a problem with the GS. See comments above. send_dtmf=in-audio This must match the entry in sip.conf (In the GS world, in-audio = inband) Sincerely, Stephen R. Besch ___ 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: Asterisk SIP + Grandstream 100 + sip.conf phone HELP
--snip-- I am having trouble setting the /etc/asterisk/sip.conf file. This is my file: 1) Add in the [general] section: disallow=all allow=ulaw allow=alaw allow=any other codec that you want to (or can) support. While some have found that this must be specified for each and every phone, I have found that it works fine specified just once in the general section. [243075] type = friend context = default secret = gol host = dynamic callerid = fono75 243075 2) Include dtfmmode=info or inband and match to phone's setting 3) I may have been too tired at the time, but once I tried using long extensions (more than 5 digits) and could not make them work either - same error you are getting. I would limit your extensions to 4 digits and see if it helps. 4) You may also need to add canreinvite=no to each phone definition. and our SIP phones configuration are the following: SIP Server: 192.168.0.102 Outbound Proxy: Empty 5) I would set this to be the same as the server if you want to make outbound calls. Hope this helps Stephen R. Besch ___ 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: RTP Read error: Resource temporarily unavailable (DTMF Issues)
Thomas Gallaway wrote: Hi I am working on this since a while now and seem to be stuck. Here is my issue: I have a bunch of Budge Tone 101's. Asterisk is set up. 4 Incoming PSTN lines. It all works fine just the DTMF is not working. I am not beind a NAT so the phones can talk directly to the asterisk server. When I want to check voicemail for example the voicemail box is 113 and password is : -- Playing 'vm-password' (language 'en') Mar 16 10:24:16 WARNING[770065]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:24:16 WARNING[770065]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:24:17 WARNING[770065]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:24:18 WARNING[770065]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable -- Incorrect password '1' for user '1113' (context = any) -- Playing 'vm-incorrect' (language 'en') -- Playing 'vm-password' (language 'en') Mar 16 10:20:42 WARNING[753681]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:20:43 WARNING[753681]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:20:44 WARNING[753681]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable -- Incorrect password '11' for user '1133' (context = any) -- Playing 'vm-incorrect' (language 'en') -- -- Playing 'vm-login' (language 'en') Mar 16 10:32:00 WARNING[786449]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:32:00 WARNING[786449]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:32:00 WARNING[786449]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable -- Playing 'vm-password' (language 'en') Mar 16 10:32:04 WARNING[786449]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable -- Incorrect password '' for user '1133' (context = any) -- Playing 'vm-incorrect' (language 'en') and so on. Maybe 1 out of 20 tries I can get the password right. The same happens with Meetme. If I tell meetme to use a PIN it wont detect it right. My sip.conf looks like this for one phone: [113] username=113 type=friend host=dynamic disallow=all allow=alaw allow=ulaw context=intern secret= mailbox=113 dtmfmode=rfc2833 nat=0 My budgetone is set to send DTMF via via RTP (RFC2833) with a payload type of 101. (tried 100 and 102) The first 2 codecs are set to PCMU and PCMA (tried to switch those arround too). Any help very appreciated! Thanks, Thomas If I recall correctly, this issue has come up many times in this list. Try not using rfc2833. Use INFO or INBAND. Stephen R. Besch ___ 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: Strange Problem
Asterisk Learner wrote: I am experiencing a strange problem and wanted to know if someone has faced any similar issues or could provide me with a way to counter this problem. I am in the process of experimenting with asterisk and trying to setup a basic functional system. I have one TDM400P (single port) and one X100P. I am using one analog phone connected to the TDM400P and I also have a couple of Xlite SIP phones configured. I can make calls out to the PSTN and I can also receive calls. The problem happens when someone from the outside (PSTN) calls the Asterisk box. I have asterisk configured to forward the call to Zap/2 (analog phone). Zap/2 rings and I can talk to the person on the other end but if I hang up first then the other end does not see as the call being hung up. Asterisk CLI shows that Zap/1-1 (FXO) hungup but for some reason the other end thinks that the call is still up and does not disconnect unless the person hangs up himself. The confusing part is that if I initiate the call then this problem does not happen. Can someone tell me what is happening and how to resolve this issue? Thanks On analog lines, the central ofice switch will signal that the remote end hung up the call, if I recall correctly, either by dropping loop current or using a brief polarity reversal. There may be other means as well. You need to make sure that you are using the correct settings to sense this properly. The reason you don't see this when you place the call is that, obviously, your end has no trouble determining when you have hung up, regardless of setting that affect the detection of the remote end hanging up. Stephen R. Besch ___ 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: Have Voice Mail tell the extension?
Zot O'Connor wrote: Is there an easy way to make the voicemail system say the extension number after the directory find (via name)? People want to know the extension once they have found the person to speed up the process. Thanks! I know it's somewhat lame, and requires more management when extensions change, but the simplest solution is to instruct users to include the extension number when they record their name for the directory. Stephen R. Besch ___ 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] Voicemail: Does the numbering of files follow file age?
I have noticed that the voicemail app always keeps filenames in in a strict numerical sequence, obviously renaming files whenever a message is deleted. I assume that message processing depends upon this sequencing. Does the file age make any difference in determining the numerical ordering (i.e., do the oldest files have the lowest numbers). While this should be true by default in the INBOX folder, it would not necessarily be true in any of the other folders, unless they are explicitly sorted and renamed every time another new message is copied into the folder. In short, are the files sorted into chronological order when they are renamed? Stephen R. Besch ___ 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: flash button on GS101
dkwok wrote: Has anyone using the flash button on GS101 to access call waiting? My experience is that it does not work. I read in the list that it may need to tweak the flash duration to under 100msec. Has anyone have any solution? Make sure that you don't have the Send Flash Event option set to YES in the GS configuration. If you do, flash will not be sent as a SIP event and the flash button won't work. Stephen R. Besch, Ph.D. ___ 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: Hanging GS101 in a upright position
dkwok wrote: Has anyone tried to hang GS101 phones on a wall? It has recess holes at the back of the base where you can hang it on a wall. What it lacks is that the handset is not supported for this upright position. Has anyone done any modification on it? I was thinking about velco the handset. On my phones, there is already a notch on the handset. Only the appropriate bump on the cradle is missing. A strategically placed 2-56 socket head screw solves the problem nicely. Stephen R. Besch ___ 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: Caller ID Name Display
htguy wrote: Got my test box up and running. Celeron 2Ghz 1Gb Ram 2x32Gb Hdd, X100P card 2xGrandStream phones. I've got everything working calls come in get processed, delivered to the extension... But what I can't figure out is how to pass the CallerID Name instead of the CallerID number? I've been banging my head, my eyes are blurry, and Google isn't answering... Could someone just point me in the right direction. Thanks -Art If you're using the GS101, save your Excedrin, it doesn't display alpha. The number is all you will ever get. Stephen R. Besch ___ 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] Supervised transfer (almost) with GS phone
I have now tested a (previously suggested) method for doing supervised transfers using the Grandstream SIP phone. It isn't perfect, but it works and is very functional. Here are the steps: 1) A call comes in that you want to transfer 2) Flash once. This switches to the 2nd line and puts the caller on hold. 3) Call the party to which you want to transfer the call. 4) Arrange the transfer and have the person to which the call is being transferred hang up. Incidentally, while it may be obvious, it is still worth mentioning that before they hang up, you can switch back and forth between the two parties an indefinate number of times. 5) WITHOUT HANGING UP, press the flash button again. This switches you back to the caller. 6) Inform them of the transfer. 7) Press the transfer button and enter the transfer phone number. (This starts a simple blind transfer). 8) Hang up to complete the blind transfer. Relative to a true supervised transfer, the only things I believe to be missing are the inability to transfer the call while the desired recipient is on the line, and the ability to pick the call back up if the transfer fails - and these may be major problems for some applications. However, it comes close enough for many purposes. For reference, I'm using a slightly modified version of the stdexten macro in the dialplan (no t or T) and either SIP INFO or INBAND on the phones. Stephen R. Besch ___ 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: Grandstream transfer into outer space
Olle E. Johansson wrote: Going back to the subject, what does the grandstream really do, SIP-wise, when you press the transfer button? Olle, The following is an exact transcription of the description given in the BT101 manual for Blind Transfers: 4.3.7 Call Transfer The user can transfer an active call to a third phone by using the Transfer button. The sequence is like this: The user presses the Transfer button and if the other voice channel is available (i.e., there is no other active conversation besides the current one), he/she will hear a dial tone. He/She can then dial the 3rd phone and then hangs up his own phone. 2 kinds of blind call transfers are supported: using REFER and using BYE/Also. The SIP message flow based on SIP REFER method looks something like this: Call Flow Diagram For Blind Call Transfer: From Transferee to Transferor INVITE - -100/180/200 ACK - - RTP Media - - REFER 202 - NOTIFY - - 200 - BYE 200 - From Transferee to Recipient INVITE - - 100/180/200 ACK - - RTP Media - The SIP message flow based on BYE/Also method looks something like this: From Transferee to Transferor INVITE - - 100/180/200 ACK - - RTP Media - - REFER 501 Not Implemented - - BYE with Also: 200 - From Transferee to Recipient INVITE - - 100/180/200 ACK - - RTP Media - I have no idea if this is accurate, I just copied it and replaced the arrows indicating direction with - and -. You can download the manual itself from the GS web site. Stephen R. Besch ___ 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: Grandstream transfer into outer space
Olle E. Johansson wrote: Stephen R. Besch wrote: Olle E. Johansson wrote: Going back to the subject, what does the grandstream really do, SIP-wise, when you press the transfer button? 4.3.7 Call Transfer The user can transfer an active call to a third phone by using the Transfer button. The sequence is like this: The user presses the Transfer button and if the other voice channel is available (i.e., there is no other active conversation besides the current one), he/she will hear a dial tone. He/She can then dial the 3rd phone and then hangs up his own phone. 2 kinds of blind call transfers are supported: using REFER and using BYE/Also. The SIP message flow based on SIP REFER method looks something like this: Call Flow Diagram For Blind Call Transfer: From Transferee to Transferor INVITE - -100/180/200 ACK - - RTP Media - - REFER 202 - NOTIFY - - 200 - BYE 200 - From Transferee to Recipient INVITE - - 100/180/200 ACK - - RTP Media - The SIP message flow based on BYE/Also method looks something like this: From Transferee to Transferor INVITE - - 100/180/200 ACK - - RTP Media - - REFER 501 Not Implemented - - BYE with Also: 200 - From Transferee to Recipient INVITE - - 100/180/200 ACK - - RTP Media - I have no idea if this is accurate, I just copied it and replaced the arrows indicating direction with - and -. You can download the manual itself from the GS web site. I'll do that. Does Asterisk work with this transfer button or not? We have implementation of both REFER and BYE/also in the sip channel. In my setting, yes it does. I've been thinking of trying transfers like these on the GS: 1) hit flash to get second line, 2) dial party to transfer to, 3) arrange transfer and have them hang up, 4) flash back to the original caller and do a blind transfer. Still, this does not fix the problem with the outer space transfers. Stephen R. Besch. ___ 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: Not Woodpeckers
Jose Quinteiro wrote: I live at sea level, and have never seen a woodpecker going at any telco equipment, but have a 60Hz hum on my POTS line through my Adtran 750. It goes away if I pick up the telephone I have cross-connected on the same line. Could it be the same problem (i.e., tip-ring imbalance?) Thanks, Jose. Lots of phones used to have a separate (3rd) ground wire delivered to the phone which was used in party line set ups. When very young, I discovered that connecting a phone tip to ground or ring to ground would result in a fabulous hum. I suspect that you have inadvertently connected one of the phones tip to ground or ring to ground. Stephen R. Besch ___ 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: Speech between Grandstream phones sounds like talking under water
Greg Boehnlein wrote: On Tue, 17 Feb 2004, Rana Dutt wrote: I was able to solve the audio quality problem by going to www.grandstream.com/BETATEST and downloading the latest beta firmware, version 1.0.4.46. I wish their Beta Releases actually had a file that showed what changes/fixes/updates have been made to the firmware. It's kind of like the blind leading the blind. :) Ironically, I suggested exactly that to the Grandstream folks over 6 months ago. They said that it was such a good idea that they would send it to their engineering department for implementation. I wonder what ever happened? Stephen R. Besch ___ 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: X100P / Echo / ZTMONITOR CAN2,3, etc.
Andrew Kohlsmith wrote: If you're getting echo of your own voice, but the remote is getting a clear signal, then Asterisk echo cancellation is working properly. It is the remote provider not echo cancelling properly. I don't buy it. If that were the case then why would I not _also_ get my own voice echoed with a regular phone plugged in to the same POTS line? And you shouldn't! Theory holds that exactly the opposite is true. With no echo cancellation at all and with an unbalanced hybrid, the local party hears the echo and the remote party does not. This is because the local hybrid listens to itself talking to the pots line and sends what it hears back to the local listener. If the hybrid is perfectly balanced, it will subtract out what it is saying to the pots line before it returns the signal to the local caller. The problem is that it rarely does a very good job of this. Neither do many commercial channel banks. There are a few common hardware glitches that can make echo a lot worse, such as a shorted leg or corroded connection in the PSTN wiring, having a hard wired extension connected to the same line or simply having a lousy CO switch at the other end. The first 2 the phone provider should diagnose and fix for you. The strange thing here is that asterisk is not removing the echo. I did notice that the 0.7.1 tar did not do the echo cancel very well and Mark suggested that I go back to the CVS, which did wonders. You might also verify that echo cancellation is actually turned on. Enter zap show channel x at the CLI, where x is one of your defined zap channels. If it's enabled, somewhere in the output you should see Echo Cancellation: xx taps, Currently On/Off. The On/Off will change from Off to On when a call is bridged. If it is not enabled, check the definitions in zapata and in your make file. ___ 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: Double digits seen using Grandstream phones
Rana Dutt wrote: My attempts to use voice mail from my Grandstream Budgetone 101 phone always fail because Asterisk is seeing either double digits or dropped digits, no matter what dtmfmode setting I try. Here is what happens for each mode: dtmfmode=info, phone set to send INFO 101: every digit is seen double, e.g. 123 is seen as 112233 dtmfmode=rfc2833, phone set to RFC2833: some digits are doubled in random places, e.g., 123 is sometimes seen as 1123, sometimes as 1233 dtmfmode=inband, phone set to send in-audio: some digits are dropped, e.g., 123 becomes 23 or 12 I have upgraded the firmware in my phones to 1.0.4.46 and Asterisk is a CVS from 2/1/04. The codec in the phones is set to PCMU (ulaw) as first preference, and the sip.conf has disallow=all followed by allow=ulaw for each phone. Early dial is turned off. Can anyone help me? Thanks, -Ron Many of us have had these problems at one time or another with GS and the solutions have been more or less the ones you've tried. In my experience, either inband or info works, providing that everything is configured correctly. The only suggestion I have is that you double and triple check that the entries in sip.conf and those on the phone are in perfect sync before testing. That is, make sure that dtmfmode= matches the setting on the phone, etc. Before you give up on any combination, reboot the phone and restart * (don't just reload). For reference, here is a typical entry from my sip.conf: [exten100] type=friend context=international callerid=Nattily Dressed 888- 100 username=Natty host=192.168.0.100 ;or dynamic, as required dtmfmode=inband ;or rfc2833 or info, as required secret=Illnevertell qualify=5000 mailbox=100 canreinvite=no ;As long as the phones are NAT'ed By the way, contrary to many posts on this list, I have disallow=all allow=ulaw allow=alaw appearing only once in the general section and all (22) of my phones and HT-286's work perfectly - just another one of those great * mysteries I guess. Stephen R. Besch ___ 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: GS BT-100 echo
Tim Sailer wrote: I picked up a GS 100 phone based on the overall good response I've heard of these phones. One thing I'm fighting with, which I can't find any info on, is a *real* bad local echo on the GS. The remote end doesn't hear it, and all the docs I see about echocancel deal with hardwired phones/ports (fxs/fso). Phone software is: Software Version: Program--1.0.4.45Bootloader--1.0.0.13HTML--1.0.0.20 if that matters. sip.conf for the phone is: [gs1] type=friend username=gs1 secret= host=dynamic canreinvite=no nat=yes qualify=1000 disallow=all allow=alaw allow=ulaw Tim Tim, Look harder in the mailing list and at the WIKI. There are literaly hundreds, if not thousands, of posts on this exact issue. Stephen R. Besch ___ 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: Annoying Beeps
Shawn L. Djernes wrote: Do you here the beeps on the phone or on the Console machine. For about the last 2 weeks I have been hearing random beeps on either of my two sip phones. I do not have a console running anywhere so I have no text printing. No, they were definately on the console, problem now fixed. Stephen R. Besch ___ 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] Annoying Beeps
Every once and a while * throws a new wrinkle at me. It has started, all on its own, to make these annoying little beeps evey time a message prints at the CLI. If I bring down * and restart, they go away for a time, then seem to spontaneously reappear sometime later. It's almost as if * is starting to experience the Terrible Twos! No one else seems to be complaining about this, but I nevertheless assume that I can somehow disable this feature, I just can't seem to find out how. Maybe something like CLI stop beeping damit? Stephen R. Besch ___ 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: Annoying Beeps
Steven Critchfield wrote: On Fri, 2004-02-06 at 13:46, Stephen R. Besch wrote: Every once and a while * throws a new wrinkle at me. It has started, all on its own, to make these annoying little beeps evey time a message prints at the CLI. If I bring down * and restart, they go away for a time, then seem to spontaneously reappear sometime later. It's almost as if * is starting to experience the Terrible Twos! No one else seems to be complaining about this, but I nevertheless assume that I can somehow disable this feature, I just can't seem to find out how. Maybe something like CLI stop beeping damit? I think that is due to there being a character on the CLI. Try hitting enter to clear the line, or hit ctrl-l to do a screen redraw and see whats on the line. That was it. I even found out how the characters (//) get typed. I have a KVM switch to pop between systems and I also have an annoying habit of hitting // rather than ctrlctrl to switch screens. It's amazing the trouble that a little sloppy typing can get one into! Thanks Steven. Stephen R. Besch ___ 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: Boards falling out...
Ejay Hire wrote: Hi. Low Temp Hot glue is what I use on my robots. Stay away from silicone (conductive) and rtv (peels traces off cheap pcb's) The only silicones that are electrically conductive are those that are loaded with some conductive material (like silver). These are rather esoteric and it is very unlikely that you would encounter them. On the other hand, many silicones are rather thermally conductive. Perhaps this is the source of this misconception. Also, Room Temperature Vulcanizing Silicones (RTV's) are, also, well, obviously, silicones. If removed with care, they should not lift traces, although, again obviously, too heavy a hand will remove traces from paper-phenolic (those brown ugly things) circuit boards. It would be really hard to lift them from epoxy-glass material (usually green). Nevertheless, hot melt glue is a fine choice. It hardens quickly, is easy to apply and is electrically and chemically inert. Stephen R. Besch ___ 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: iax, trunking, etc.
Chris Clifton wrote: The majority of sip to pstn gateway providers (vonage, voicepulse, and others) appear to be setup for a one line only type of set up. Their web sites seem to be heavily geared for these one line setups. Anyone willing to comment on what type of pricing plans these providers offer when using iax2 trunking or other methods with asterisk to send multiple (and possibly simultaneous) calls through their gateways ? Voicepulse told me that there was no additional charge to enable trunking. Stephen R. Besch ___ 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: Anyone used a Grandstream ATA286 with Asterisk
MLS Drop for SysAdmin wrote: an associate of mine sent me an email of the slick sheet on this one. I understand that mentioning this vendor has resulted in some flamethrowing on the list, and I do not want to cause trouble - just looking for some info. Thanks! Sam Z I have one in service. It has no serious problems. The one issue we have had is that it adds an additional 2-wire to 4-wire hybrid and therefore adds an echo source that the remote end hears (varies depending upon what you plug into it) - and, I suspect that, because of the somewhat longer packet delay, the echo from this source is not well corrected by the * echocanceller. Stephen R. Besch ___ 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: Boards falling out...
Greg Kedrovsky wrote: I have a TDM40B, 4-port fxs card. Each port seems to have it's own little board on the fxs card. Each little board is not sodered in, but rather hangs (I have a vertical case for the server) on what I would call jumper pins (sorry, I'm not a profession geek, just a wannabe). One of my little boards, over time, slides off those jumper pins. I just noticed it this morning. I had to power down, seat it, and power up again. That's a pain. We did, though, have an earthquake this morning. That may have shaken things loose a bit. But, it wasn't much to speak of (long, but not strong). Has anyone else experienced this problem? What could I do to solve it (seat the little card a little more permanently)? Thanks ahead of time. -Greg There are three issues here, relating to the other posts on this topic. Don't use loktite. Loktite is what is called an anerobic adhesive. Specifically, it is catalyzed by contact with metal in the absence of oxygen. As such, it will only cure (in the absence of some other chemical activator) only down inside the pin sockets, holding them together. The rest will stay uncured and spread all over other stuff. This may essentially make them a single use contact. The silicone is a good bet. The acid referred to is the acetic acid (i.e., vinegar) released when the monomers in the RTV goo cross react to form the silicone. Once the cure is complete, there is no acid production and what was produced diffuses away. Mild acids are not terribly corrosive to most metals, and not at all corrosive to gold. The types of RTV that don't produce acid may actually produce alkali (ammonia), which is far more corrosive, but also diffuses away readily. Nevertheless, I would stick to the stuff that smells like vinegar. Finally, I have found that the best approach is the simplest, when it works. If you can get one of those nylon tie-wraps around the daughter card in such a way as to hold it in place, this is the best - and most reversible approach. Sometimes, there are appropriate holes in the motherboard, othertimes the ty-wrap can be snaked around under the connector - however, don't run it under any other type of component. I have even drilled holes in 2-layer circuit boards, but I would not advise this unless you really, really, really know what you are doing. Finally, if the female side of pin sockets are loose enough to let the dayghter cards fall out, they may also be the source of noisy, intermittent connections. Sockets of allmost all kinds are notorious for this kind of thing. I can't tell you how many times I have repaired a flakey circuit board by removing the sockets and soldering in all the (formerly) socketed chips. The square pin spring contacts in those connectors are only designed for a few insertion/removal cycles. If that is the case, you should get a good repair tech. to replace them. Good luck and hang in there. Stephen R. Besch ___ 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: Boards falling out...
Jon Pounder wrote: nail polish and liquid-paper work fine for this sort of stuff. Too brittle. Adhesion to metal and many plastics is marginal. Fine for places where there is no shock (of the physical kind). If this is earthquake territory, stick to the silicone or ty-wraps. Stephen R. Besch ___ 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: Compiling while * is running
Stephen R. Besch wrote: I just fetched today's cvs (1/30/04 11:10:31). Compiles/installs on my test machine (ASUS A7V, 900 MHZ). However, If I try to compile on my production machine (Elite K7S5A, 2.4GHz, 512MB) while * is running the zaptel and asterisk compiles seg fault. I am assuming that they will compile correctly if I bring down * and rmmod the zaptel driver. 0.7.1 compiled and is now running. Is there a way to safely compile while * is running, so that I can minimize down time of the server? Here's the update on the seg fault problems. After bad memory was suggested, I checked to see what memory the tech who assembled the machine had used. I don't know why, but the module was not in the recommended list for the MOBO at Crucial.com. I ordered one of the recommended modules. It arrived this morning. I installed it and all of the seg faults are gone. I've learned my lesson. Next time, I will assemble my own machine! Thanks again for all the help. Stephen R. Besch ___ 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: Compiling while * is running
William Waites wrote: While your problem is most likely bad RAM as other replies have suggested, there is another thing to keep in mind. Some implementations of dynamic module loading have problems if a loaded module is overwritten on the disk. What this means is that it is safest to stop Asterisk just before running make install, else the running instance may mysteriously segfault at that point. /w Well, all the subsequent messages on this topic notwithstanding, it still seg faults when * is not running. I have ordered some new memory. I'll post the result. Stephne R. Besch ___ 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: Question on setting up asterisk with hunting lines
[EMAIL PROTECTED] wrote:
*My apologies if this message is posted 3 times, I was trying to sent it to
the list once before I am a list-member, the second time before I was
approved.
Can anyone point me to some resources on using hunting lines with Asterisk?
Sales support of my telco have no idea what I am trying to do. They asked
what pbx system I am using, I was like Aster... never mind =)
I am trying to setup asterisk to take in 5 hunting lines.
Where one phone number would get published as our companies main IVR entry
point, and the calls will get distributed into the Asterisk system internal
extensions via the 5 available hunting lines.
I am lost here.
When a customer dials the main number, does it (A) get call transferred to
an available channel by a dial plan with asterisk, or (B) the telco
automatically checks to see if the main number is busy and transfer to the
next hunting line?
On incoming, telco does the hunt and rings an open line. You need 5
FXO's to handle the lines. Then set them up in a group. I use an ADTRAN
channel bank with a digium T1 card and have the following in
\etc\asterisk\zapata.conf, for 5 lines:
[channels]
:
:
context=default ; or whatever
:
:
signalling=fxs_ks
group=1
channel = 1-5 ;Group1
The fxs-ks is not a mistake. When you talk to an fxo (central office
line), you have to look like an fxs (a telephone). These five lines will
then drop into the default context in your dialplan. For a channel bank,
you also have to define the T1 channel assignments in \etc\zaptel.conf.
Again, in my case:
fxsks=1-5
fxoks=6-24
You will have to adjust these for your hardware and mapping assignments.
For outgoing hunting, just use group1 in your dial statements, like this:
[globals]
PSTNTRUNK=Zap/g1;I do this for convenience/flexibility
[some context or other]
exten = _pattern,1,Dial(${PSTNTRUNK}/${EXTEN})
If (A), do I flash, dial to the available hunting line with my dial plan,
and disconnect the original call (similar to a 3 way call conference).
Would this even work on a external telco line?
If (B), this would be simple, I would assume Asterisk can listen to all 5
fxo and run the same IVR script
Here is my setup.
5 FXO hunting lines/19 FXS analog phone goes to a channel bank, then to a
Digium T1 card.
Thanks in advance,
Sam
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[Asterisk-Users] Re: Grandstream Firmware ?
Greg Boehnlein wrote: On Thu, 29 Jan 2004, Michael Welter wrote: I have 1.0.4.45 (beta) on my tftp server. Try it at 66.250.23.58. Cheers, Michael Welter Is there a changelog available for the Beta release train? I'm looking to see if they have fixed Early Dial yet. When GS connected to my * server to examine the problem, they promised to keep me posted on the early dial problem. I haven't heard anything yet, so I am assuming that it has not been fixed. ___ 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] Compiling while * is running
I just fetched today's cvs (1/30/04 11:10:31). Compiles/installs on my test machine (ASUS A7V, 900 MHZ). However, If I try to compile on my production machine (Elite K7S5A, 2.4GHz, 512MB) while * is running the zaptel and asterisk compiles seg fault. I am assuming that they will compile correctly if I bring down * and rmmod the zaptel driver. 0.7.1 compiled and is now running. Is there a way to safely compile while * is running, so that I can minimize down time of the server? ___ 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
