Re: [Asterisk-Users] Asterisk and Linejacks
I actually have a phonejack, not a linejack. So I probably can't help. What does the linejack config file look like? You need a context specified (like 'default') that has a 's' extension (aka start) that answers the call. Something like: [interfaces] context=default mode=fxo format=g723.1 device = /dev/phone0 -Andrew greg wrote: I found a message from you to the asterisk users mailing list from 2001. I was wondering if you got (or still have) an asterisk system working with the linejack? If so, would you be willing to assist me with mine? I seem to have things working, and * says that caller ID is coming in, but I can't get * to actually answer the call. Thanks, Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Crisco Softphone
Derek Bruce wrote: depends on which Cisco softphone you are refering to... they have a few different versions... including an NBX version which will not work with Asterisk... - Original Message - From: Tim Sailer [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Tuesday, March 16, 2004 2:57 PM Subject: [Asterisk-Users] Crisco Softphone I was given an eval of the Cisco softphone to try out. Has anyone gotten this to work with * yet? Tim From the subject I would have guessed some sort of vegetable shortening based phone, not an NBX one. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
James Sizemore wrote: exten = 6500,1,Answer exten = 6500,2,Wait,1 exten = 6500,3,VoicemailMain2 Or should I say, Me too! Is this the bug for the case in question? CSCed48311: Media takes 0.4 sec to be set up Thanks. -Andrew Yes the problem is that when making outgoing calls, there is enough of a delay in the call setup once the remote side picks up, that people that answer the phone hello will be heard saying o or if they talk fast enough not heard at all therefor leaving a very awkward silence at the start of a call. According to the bug release notes this is caused by the DSP setup on the 7960. I would guess that it must need to setup the correct codec once it is selected and that takes some time (400ms apparently). Perhaps they could create a 'leave the dsp setup for codec X and never change codecs' config option. :-) This is very annoying. A earlier person suggested answering the calls before dialing and playing a ringing sound till the start of the voice. That may be a work around of sorts for some, you will hear a ring then a congestion tone on call that can't connect, or a ring before a operator messages (say to dial one before the number) that most users may not be used to. I'll be playing with that ideal to see what odd effect a ring has before call setup causes. The work around may be less annoying then the problem. smile I'll see. Sounds good. I have not been that bothered with it when I make a normal voice call. It is mostly annoying when hitting the messages button on the phone. My delay helped that situation. Perhaps on calls where asterisk is proxying the rtp stream we could have an option to tell asterisk to open the connection to the 7960 before the connection is setup on the other side of the call. So the 7960 gets a head start. It would force the codec but that is fine by me, my G.729 is preferred and I don't mind asterisk transcoding since I have a low number of calls. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
Steve Creel wrote: On Wed, 10 Mar 2004, John Fraizer wrote: For what it's worth, I don't have any delay between answer and audio with my asterisk server and 7960G either originating or answering. It doesn't matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's pretty much instant (not detectable by humans at least). So, there may be some truth to the fact that the delay is caused by the Asterisk install in your case. There are so many variables that it is very hard to tell but, since I don't see the delay, I am leaning towards it being an Asterisk implementation issue. Can you test this with an extension that goes into VoiceMailMain(). My 7960 and 7960G phones both get the first couple letters of Commedian Mail cut off (usually ...median Mail). Just trying to quantify the delay we're talking about... exten = 6500,1,Answer exten = 6500,2,Wait,1 exten = 6500,3,VoicemailMain2 Or should I say, Me too! Is this the bug for the case in question? CSCed48311: Media takes 0.4 sec to be set up Thanks. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Codecs [G.729]
Unavailable ID wrote: Hello all, I'm looking for advice for codec that works best for asterisk. Anyone has real testing with all codecs, specially with G.729. I have tested with single call on few codecs that come with asterisk by using IPTraf and the rate as of below: ulaw 64 Kbps, sample-based Also known as alaw/ulaw 166kbits/sec alaw 64 Kbps, sample-based Also known as alaw/ulaw 167kbits/sec gsm 13 Kbps (full rate), 20ms frame size 66kbits/sec speex 2.15 to 44.2 Kbps n/a iLBC 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size 57.6kbits/sec G.729 8 Kbps, 10ms frame sizelicense Have anyone test it with G.729? Please let me know. Thanks. Are some of these numbers for the full-duplex traffic? -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Codecs [G.729]
Andrew Gillham wrote: Unavailable ID wrote: Hello all, I'm looking for advice for codec that works best for asterisk. Anyone has real testing with all codecs, specially with G.729. I have tested with single call on few codecs that come with asterisk by using IPTraf and the rate as of below: ulaw 64 Kbps, sample-based Also known as alaw/ulaw 166kbits/sec alaw 64 Kbps, sample-based Also known as alaw/ulaw 167kbits/sec gsm 13 Kbps (full rate), 20ms frame size 66kbits/sec speex 2.15 to 44.2 Kbps n/a iLBC 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size 57.6kbits/sec G.729 8 Kbps, 10ms frame sizelicense Have anyone test it with G.729? Please let me know. Thanks. Are some of these numbers for the full-duplex traffic? Ok, my question doesn't even seem that clear to me. :-) What I mean, is that the G.711 numbers for example look like both directions *combined* so the actual rate would be more like 83Kbit/s. (much like listed on the wiki page) So for G.711 a/ulaw, gsm, iLBC etc I was wondering if that is a single direction, or the combination of both directions? -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phone with large display
Jonathan Moore wrote: I have a client that would like to purchase 12 IP phones for an office environment. We were planning to purchase the Snom 220s, but apparently they are still not available in the US. The new Sayson 480 also would fit the bill, but won't be available until April. They have looked at the Cisco 7960 also as an option. They tried and liked the Polycom, but the support was just not workable. 1) Are there any other IP phones on the market with large displays that work with *? 2) Do you know of any decent analog phones with large displays/multiline ( assuming mainly for caller ID) that could be paired with an analog to IP converter? Total package (IP phone and powewr supply or analog phone + converter) should probably cost no more than the Snom 220 at about $325-$350 per unit. You can get refurbished 7960G units in that range and new for around $399 according to a quick search on Froogle. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BCM Wireless SIP Phone
Steven Thomas wrote: *Hi,* Has anyone tried this Wireless SIP phone with Asterisk? If so, any limitations? Thanks. http://www.bcm.com.tw/product/productIS.htm This WiFi phone and the Zyxel Prestige that was just mentioned both look like the Pulver WiSIP phone. Since that works I would tend to assume they all will. I don't know if BCM is the OEM on this product or not, but since I bought an IP phone from ArrayVox that appears to be the same as the BCM HP300 I would assume they are OEMing their products. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 conference ?
Chris Clifton wrote: I would buy this, but my 7960 is using g729a on both lines that I'm dialing out on (to conference), my * installation is licensed for 3 g729 channels. What codecs are you using ? Is there a conference config in the 7960 that I'm missing ? I can make inbound and outbound calls just fine on this phone using g279 to other sip phones and the pstn all day long. Thanks, Chris The 7960 can do conferencing internally (without the aid of a SIP proxy) and I believe it can only conference with G.711 since that doesn't required DSP resources for each lines codec, just for mixing. I'm not sure if you're talking about Asterisk provided conferencing, or phone provided. -Andrew ___ 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] Debian Testing / 2.4.22 / zaptel problems.
I'm having trouble getting zaptel to work on Debian Testing (Sarge) with a 2.4.22 kernel. The errors I am seeing with 'insmod zaptel.o' are: ./zaptel.o: unresolved symbol devfs_unregister_R1c83d91a ./zaptel.o: unresolved symbol remove_wait_queue_R1bc53d4c ./zaptel.o: unresolved symbol __pollwait_R8ee1d6fc ./zaptel.o: unresolved symbol remove_proc_entry_R30e2f8d2 ./zaptel.o: unresolved symbol devfs_register_chrdev_R69b94695 ./zaptel.o: unresolved symbol proc_mkdir_R57411544 ./zaptel.o: unresolved symbol devfs_register_Rbf40312b ./zaptel.o: unresolved symbol devfs_generate_path_R0c78ae56 ./zaptel.o: unresolved symbol add_wait_queue_R498065b0 ./zaptel.o: unresolved symbol create_proc_entry_Rebcd0c7f ./zaptel.o: unresolved symbol devfs_mk_symlink_R5b830122 ./zaptel.o: unresolved symbol devfs_mk_dir_Rbf4f104c This is with a stock install, with updates, and asterisk / zaptel from cvs. Any ideas? -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debian Testing / 2.4.22 / zaptel problems.
Jeremy McNamara wrote: Typical version skewTry linking to the kernel source that is actually running on the box. Well as far as I can tell, the only version I have on the box is 2.4.22-1. I certainly only have 'kernel-headers-2.4.22-1' installed and 'linux' symlinked to that directory in /usr/src. Are you saying my /usr/include would be skewed? Since I thought that was from the libc6-dev, not really kernel related? I will try removing all of the -dev packages and re-installing them. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to use * as a gateway?
Pavel Litvinenko wrote: Joseph Finley wrote: I'm not sure if I am wording this correctly, but I'll try. I have a Cisco 2621 w/ a couple FXO and FXS ports. I have a couple cheap analog phones plugged into the FXS ports. I am able to get * to ring those phones when a call comes in, but I cannot get the phones to dial out. I guess it's all syntax that I'm doing wrong. Does someone have a couple small snip-its to accomplish this? This is what a buddy of mine uses to call my pbx extensions. ! voice service voip h323 sip bind all source-interface FastEthernet0/0 --- the public IP interface ! ! The Cisco 7960s only do these two codecs. (also g711alaw) voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! An analog port on the 3725 router. voice-port 1/0/0 description POTS Test Phone ! ! The local number for the analog port. dial-peer voice 100 pots application session destination-pattern 6110 port 1/0/0 ! ! forward anything 6XXX to my pbx at 1.2.3.4 dial-peer voice 111 voip preference 1 destination-pattern 6... voice-class codec 1 voice-class h323 1 session protocol sipv2 session target ipv4:1.2.3.4 ip qos dscp cs5 media no vad ! ! I believe this just tells where the server is, it doesn't REGISTER. sip-ua sip-server ipv4:1.2.3.4 ! Newer Cisco IOS is supposed to be able to register via SIP, but the version my buddy is running doesn't currently support it. But he is able to dial my pbx easily, and I can setup the sip.conf with a default ip for his router, etc. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk integrated with ventrilo or teamspeak
Steven Critchfield wrote: Please read past rants about the action you took to create this message. Hint: You broke the thread by replying to an unrelated thread. Could all of the thread police please just reply personally to the offending party? The amount of people interested in the rant is probably similar to the amount interested in rants about grammar, spelling, newlines, etc. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Handytone 286 - calling out
Senad Jordanovic wrote: Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just hangs in there. ATA is behind NAT, registers to an * with public IP with no problems and it uses 1.0.4.17 firmware. Web config screen has detected firewall/NAT type is open Internet as network firewall. Are you able to use tcpdump on the asterisk box to capture traffic from the ATA? Or Ethereal if you have X installed on the Linux box. It would be interesting to see if the ATA sends anything after you dial the '1234#' sequence. On my Grandstream 101 phone I have not had any trouble placing calls. I don't have the 'Outbound Proxy' field configured, and I re-ordered the codec preferences as well. Other than that it is pretty much stock with SIP server / user and authentication configured. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk] GSM access
Robert Murray wrote: Hi Mark Did you or anyone else ever find a satisfactory solution to this? Are there any phones which provide voice through the serial connection? What about the nokia card phone - does it have open source drivers? Cheers Rob On Sat, Jul 13, 2002 at 10:31:53AM -0500, Mark Spencer wrote: Does anyone (maybe in Europe) know how I could build a GSM compatible channel for Asterisk, so that one could call other mobile phones from Asterisk, or build a portable phone system, with GSM channels being used for outside access? Is there any hardware for PC's or a way to rig up a phone with a serial connection and a sound card to use it? Mark It seems to me that the Nokia 32 GSM terminal would be the best bet. Has anyone used one yet with Asterisk? -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0
Dan wrote: Hi, - Original Message - From: Patrick Cantwell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 6:48 AM Subject: RE: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0 I am having the same problem. I'm running it from the command line out of c:\iaxcomm, and it loads, but only shows up in task manager. This is a WinXP Pro box. Thanks, Pat Very very very STRANGE! It seems that I have not received this mail from Michael, even it is posted to the distribution list and this is not the only one?!?! Someone else with this problem? Well, I saw at least 4 messages recently from the asterisk-users mailing list, that spamassassin scored over 5. I believe the sender's SMTP relay was in the various open relay databases. So you may have filtered some messages if you're using a spam filter and haven't whitelisted lists.digium.com. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bad echo on outgoing calls
Andrew Joakimsen wrote: The X100P cards have horrible echo problems. I've heard talk about this being fixed, but havent seen anything done about it. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Larry D. Black Sent: Saturday, November 15, 2003 3:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Bad echo on outgoing calls I have just installed and configured asterisk I have been playing with software phones and an analog phone plugged into a TDM card. I have one line coming in on a X100P card. My X100P works quite well if I don't adjust the gain. Unfortunately it is a bit on the quiet side without the adjustment. I'll test it out with the echotraining and the gain settings. In the past with gain enabled, the echo would correct after 5-10 seconds of conversation. This is with MEC2, and I tested with and without the aggressive suppressor. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Authenticating zap callers. (callerid or PIN code)
Does anyone have any decent working examples of providing a means for a remote caller to authenticate and get internal access through the PBX? I would like to be able to call in with my cell (or any other line), enter a code and then be able to dial just like if I had picked up my IP phone at home. My initial scenario would be letting me hit a number during my initial announcement and then prompting (or just a subtle signal) for a certain code. Once the code is entered successfully I would either just get a second dial tone, or build an IVR to let me pick the function or service. A second scenario might be to just look at the callerid and if it is my cell to just throw the call right into the IVR or correct context, etc. If anyone has a pointer to sample configs for this I would appreciate it. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Authenticating zap callers. (callerid or PIN code)
Eric Wieling wrote: show application disa Thanks. I think the reason I have been having so much trouble is that my initial DTMF is not recognized and I have been hitting it again, holding it down, etc and my authentication has failed. My X100P context (now) looks basically like this: exten = s,1,Dial(SIP/[EMAIL PROTECTED],14) exten = s,2,Answer ; Answer the line ;exten = s,2,Zapateller(answer) ; Answer since nobody picked up exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Background(my-noanswer); I'm not available, etc... exten = s,6,Voicemail2,s1234 exten = s,102,Voicemail2,b1234 exten = 2,2,DISA,12345|local Turning off the Zapateller made a big difference, I can reliably get to the DISA check on the first press of '2' rather than having to press it twice typically. In either case I am pressing it once the Background() starts, not during the Zapateller(), but apparently Zapateller() is causing problems for me. Does anyone else have trouble with DTMF after using Zapateller()? -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get the g.723 codec?
Gavin Hamill wrote: On Mon, 2003-11-03 at 15:14, Eric Wieling wrote: Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html From what I remember when I looked into this about a year ago, this isn't even the end of it, since whilst DSPG represent /most/ of the IP holders on the codec, there are still others, and if you want to be completely sure of being legally in the clear, then you must reach seperate licensing arrangements with them If only some of the hard-phones would use Speex or similar, then all these problems would Go Away, and the production costs for the phones could drop, giving the manufr. the same amount of margin, but at a lower market cost. Cheers, Gavin. Asterisk doesn't seem to support SPEEX all that well. Has anyone had any luck getting it to work with X-lite? -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get the g.723 codec?
Steve Underwood wrote: Hi Thomas, Unless you have a *very* specific need to use G.723.1 for compatibility with someone else, forget it. It is pretty much an obsolete product. Licencing is also a pain, as there is not patent pool for it. G.729 is expensive to licence, but at least it is relatively strightforward. If you think you will save some bits using G.723.1 instead of G.729, think again. The saving is minute, because of the huge overheads IP imposes. Regards, Steve I was measuring about 32-36 Kbit/s for a G.729 call, and around 20-22Kbit/s for G.723. This is at the DSL router, so it includes all of the overhead. If you're on a dialup modem, that can make or break the call. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get the g.723 codec?
Brian West wrote: Asterisk doesn't seem to support SPEEX all that well. Has anyone had any luck getting it to work with X-lite? Speex works perfect with IAX but not that crack headed x-lite stuff. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Ok, so it is essentially worthless as a low bandwidth codec option to remote ip phones. Why do you use SPEEX instead of GSM? -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco or Snom ???
WipeOut wrote: Bartosz Jozwiak wrote: What is better? Cisco 7960 or Snom 200 ?? Bartosz How much do you want to spend and do you really want the name?? Haven't used the Cisco, way too pricey.. Snom's work great.. You have to look at what CODEC support you want also. My Cisco 7960 phones work great, but the only low bandwidth codec is G.729. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] looking for a place to buy SNOM or Cisco Phones (Cheap)
Todd Wallace wrote: Does anyone know where I can buy SNOM or Cisco (new or used) phones the cheapest. I need a few Todd Wallace Uh http://www.ebay.com/ -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Firmare
[EMAIL PROTECTED] wrote: Does any have a copy of the 30202 7960 firmware? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users It is on Cisco's FTP server if you have a CCO account. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Cisco 7960 Firmware Upgrade
[EMAIL PROTECTED] wrote: I'm still stuck on this. The * is on a private network IP 192.168.0.7 and the 7960 is on 192.168.0.6. When I show 'sip peers' the 7960 appears to be registered although I can't make any calls and still get the packet retries. I have also checked and re-checked the settings on the phone. Any help is appreciated. Kevin Can you send me privately (not via 'reply') your sip.conf entry for this phone? Also the output from telneting to the phone and typing: show register show status show flash show dialplan show config You can remove the private info or not, up to you, but send it to my email not the list. :) I have several 7960 phones and one 7960G working great with 4.4 5.3 firmware. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog FXO Card
Mark Spencer wrote: For the record, you can turn on the speaker by doing monitor=1 when you modprobe wcfxo, e.g.: # modprobe wcfxo monitor=1 Sweet, this is good to know! It would be nice to have a feature that kicks the monitor on for the first 30-60 seconds of a call or something. It would make it easier to hear the voicemail message that some is leaving you. :) As for the remainder of the discussion of potential clone hardware, let me keep my statements very brief and to the point. When you purchase a product from Digium, you're getting more than a piece of hardware. First, when you buy from us, you know it's going to work. All we do is Asterisk, and when we sell any piece of hardware, it *will* work with Asterisk and you *will* get the support your need (including replacement should there be a problem). Looking at the eBay auction you're saving like $40, but not getting the officially supported card. $40 (more) for great support is priceless. Where else do you get direct access to the *core developers* for $99? I think you could spend $99,000 with big company xyz and not get anywhere near somebody who *has a clue* about what is really going on inside the pbx. How often do you imagine a key Nortel developer offers to ssh in and *fix* the problem right then and there? -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco IP Phone 7905G
Andrew Gillham wrote: [EMAIL PROTECTED] wrote: Has anyone had any success using a Cisco 7905G phone with Asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Mine works terrific. My 7940 (non-G) phones work nicely as well. What kind of issues are you having? -Andrew Ok, I was confused here, I read '7960G' not 7905G. Sorry about that. :) -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] disconnect when 7960 far from * (was Re: Pointer to upgrade 7960sip beyond v3.2.0?)
Louis-David Mitterrand wrote: On Thu, Sep 04, 2003 at 10:56:10PM -0700, Andrew Gillham wrote: Unless you're hoping to load Linux or some pirate image in the future, there is no reason to stay with the old code. At least I have not experienced any new issues I can attribute to the update to 5.3 code. Hello, I bought my 7960 phones used with the 4.4 sip image and suffer from disconnections after 3/5 seconds if the phone is connected to a remote asterisk, for example at the remote end of a VPN (when the 7960 is on the same LAN as asterisk all is well). Do you think upgrading to 5.x series images would solve that issue? Thanks, Well, I have two people using 7960s remotely, both at least 120ms away and have never seen this issue. They are using 4.4 currently. Since I haven't seen the issue with 4.4, I can't guess whether 5.3 fixes it. What settings are you using in /etc/asterisk/sip.conf for these phones? For example I have: [1234] callerid=Person 1234 1234 context=internal type=friend secret=pass host=dynamic mailbox=1234 qualify=5000 nat=yes ;canreinvite=yes Have you tested these phones with Pulver Free World Dialup? (just to confirm the issue is with Asterisk only) -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960
Sorry for the late reply on this.. Ben Wern wrote: Andrew, Thanks for your help! I did have the outgoing proxy set -- since I had FWD set up on line 1. I removed all the FWD stuff, and the outgoing proxy. I altered the entry to have the qualify, canreinvite, and nat lines and also altered the user id to be a number. Now I'm able to call other local extensions, but I can't call into the Cisco. But it's progress! The outgoing proxy apparently overrides the per line proxy, so you want to leave it empty and just configure each line with the appropriate proxy. I can also call out to FWD, but audio drops after a few seconds. Don't even want to think about getting FWD calls back into the network. exten = 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) This didn't work - what does the @1000 indicate? If your sip.conf entry is under '[1000]' and you have the Cisco configured as '1000', this is tell the 7960 which line is ringing. That is why you get the 404 not found because it doesn't know which line you're calling. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco IP Phone 7905G
[EMAIL PROTECTED] wrote: Has anyone had any success using a Cisco 7905G phone with Asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Mine works terrific. My 7940 (non-G) phones work nicely as well. What kind of issues are you having? -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pointer to upgrade 7960sip beyond v3.2.0?
Joseph Finley wrote: Rich, you can do **# and go into the Network config and hit **# again, you should notice the LockPad come unlocked and then you can make changes. If you upgraded, the default password is cisco Joe This key sequence does nothing in the newer SIP code. (if it ever worked on SIP) You have to go to the menu item to unlock. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Arraycom voip phone
Paulo Mannheimer wrote: Hi All, Does anyone have any experience with the ArrayCom VoIP phone? I bought one a couple of weeks ago, it used to work quite well with * until I misconfigured one option. I now cannot make it work anymore, because the phone boots up, doesn't find a valid SIP gateway, resets itself and keeps rebooting indefinetely ;-( Their technical support refuses to answer my questions. Any hint on a master reset? PauloHM I don't know about any Arraycom, but I have an Arrayvox that did something like this, but I was able to reconfigure it by not having it plugged into the network on bootup. You could give it a shot. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sample configs
Travis Johnson wrote:
Hi,
Ok, the phones are working and seem to be loading the correct info
from the tftp server. However, I am unable to make them perform any
functions (calling another extension, going to voicemail, etc.). I do
not have any telephony interface installed yet, only a single ethernet
card. Do I need to install the ztdummy driver to make any of this
work? And if so, how do I do that?
Thanks,
Travis
Microserv
My config stuff was just for the Cisco 7960 part of it. You will still
need to configure extensions and such in Asterisk.
I usually do something like this, assuming sip.conf is using [1000] and
the context is set to local for your SIP phone.
/etc/asterisk/extensions.conf:
At the top:
PHONE1000=SIP/[EMAIL PROTECTED]
Later in the file:
[local]
exten = 1000,1,Dial(${PHONE1000},20)
This let's me easily change where extension 1000 rings. The
SIP/[EMAIL PROTECTED] just tells Asterisk to call line '1000' at the device that
is in sip.conf under [1000], which is needed on a Cisco IP phone,
particularly if you have multiple lines configured.
-Andrew
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between Cisco 7940/7940G, 7960/7960G
Mike Ciholas wrote: I'm shopping for good deals on Cisco phones. Forgive my ignorance, but I spent over an hour at Cisco's web site and Google trying to find a definitive statement as to the differences between a 7940 and 7940G phone. Anybody know? Should I prefer the G and why? Related question, is the 7960 worth so much more than the 7940? Has only 4 more buttons that I can see. Anything else under the hood that makes it worth that? The 'G' is Global. They redesigned the buttons on the right to be icons instead of the English words help/settings/messages/... so a template could be stuck over those keys in whatever local language. I would imagine there might be some hardware updates as well, but it is essentially the same phone as the 7960. Personally I have been buying used 7960 phones on eBay and I don't really see a reason to buy the 7940. :) -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960
Andrew Joakimsen wrote: exten = 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) This didn't work - what does the @1000 indicate? It shouldn't be there, If it's defined as 1000 in sip.conf make your dial string exten = 1000,1,Dial(SIP/1000,20,Ttr) You need 'SIP/[EMAIL PROTECTED]' if you want to tell the Cisco what line you are calling! This just says I am calling the line configured as '1000' on the Cisco device that is defined as [1000] in sip.conf. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pointer to upgrade 7960sip beyond v3.2.0?
Rich Adamson wrote: I've can now get to the net sip configuration panel, and I've got SIPDefault.cnt file that gets loaded at boot time. One of two 7960's is upgraded to v4.4, however the second one constantly reboots (only with v4.4) and never gets to a usable status. Using a sniffer trace to observe, the reboot happens after dhcp is proper and SIPdefault is read by the phone (via tftp). Within milliseconds of reading that file, the phone reboots, then repeats the same process. The first phone boots v4.4 correctly every time. Check the 'Boot Load ID' under settings, 5, 3. They are probably different on the two phones. A 7960 of mine running 4.4 has PC030300, while a 7960G running 5.3 has PC030301. I believe the boot code should get updated once you cleanly boot the 4.4 code, but potentially you have to load 3.2.2, then 3.2.3, then 4.4.0 to get updated correctly. Also, telnet in and do 'show flash' to see what you have loaded there. Normally you'll have two copies of the current working code, APP1 and APP2, plus DSP code. The output of 'show config' might be useful to look at as well. I had a similar problem with having to load a stripped SIPDefault.cnf, but I needed to load 3.2.2 on my phone first before upgrading from the old skinny image that was on it. Basically the phone kept rebooting because it would load the image, but it failed the internal version check because the old boot code couldn't comprehend the image names with '-' characters. That is why it was necessary to boot 3.2.2 first as it was apparently the last code that could be booted by the old boot code, but was able to update it and load the newer code. Late yesterday, I removed the statements in SIPdefault after comments relative to ...added in Release 3.1 and the unstable phone will bootup and stay working on v4.4. If I add those config statements, the constant reboot happens again. I'm not sure what is going on here, since older boot code shouldn't really be able to load the 4.4 and run, but you might be caught in between usuable boot codes. The old SCCP code - SIP 3.2.2 - SIP 3.2.3 - SIP 4.4 worked ok for me. Let me know via private email (not just 'reply') if you still need help. Is there an administrators or diagnostic manual fot the 7960sip, or has cisco left it up to all of us to figure out how to diagnose strange problems like this? There are more details available, but you may need to have a CCO account to access the documents. I'm somewhat reluctant to try v5.4 code given the warning that one can't go back. Unless you're hoping to load Linux or some pirate image in the future, there is no reason to stay with the old code. At least I have not experienced any new issues I can attribute to the update to 5.3 code. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960
Ben Wern wrote: I'm trying to get my Cisco 7960 configured to work with Asterisk, with no luck. I'm sure I'm missing something very easy... since I know others have this working. I've stepped through Andy Powell's excellent Getting Started with Asterisk, and it works for my X-Lite softphone. My sip.conf entry for the cisco looks like this: [cisco] type=friend username=cisco secret=1234 host=dynamic defaultip=[The IP of the 7960] mailbox= context=sip callerid=Ben 1 Use something like: [1000] callerid=Ben 1000 context=sip type=friend secret=1234 host=dynamic defaultip=youraddress mailbox=1000 Optionally: qualify=500 canreinvite=no nat=yes And the related extensions.conf entry: exten = 1,1,Dial(SIP/cisco,20,tr) You might want this to be: exten = 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) The Cisco config itself.. Line 1 is set for FWD. Line 2 is: Make sure you didn't set the 'outbound proxy' setting on the phone, that will force everything to the proxy. Name: cisco Shortname: cisco Authentication Name: cisco Authentication Password: 1234 Display Name: cisco proxy address: [The IP of my Asterisk installation] proxy port: 5060 Set all the 'cisco' entries to '1000' in this case. I have several 7960s working with Asterisk, so I can help you out more if you need it. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does Asterisk handle connecting two IP end points?
On Thu, Jul 03, 2003 at 03:57:09PM +, WipeOut . wrote: IIRC asterisk by default will not participate in the call between two SIP phones.. It will help establish the session to the correct UA and then have nothing more to do with it unless the call is transferred to another UA in which case Asrerisk will again be involved in setting up the call.. Asterisk will be handling the signalling for the call, not the voice stream. If you look at 'show channels' or 'sip show channels' while the call is up you will see that Asterisk is aware of it. Running tcpdump will show you that the phones are still talking to Asterisk. So no when 2 SIP UA'a are connected there should be no CPU load on the Asterisk server.. Well there is a minimum amount just for keeping track of the call, but per call is very low. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Telephone Tree
On Wed, Jun 11, 2003 at 07:44:47PM -0600, Dylan VanHerpen wrote: Well, I guess you'd have to include a disclaimer not to use it for marketing or political purposes ;) Perhaps an 'abuse' clause is needed. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
