Re: [Asterisk-Users] Grandstream GXP-2000
One thing to keep in mind with some of these phones is that physical quality can not be changed easily, why software is a different matter, though no guarantees. I have been working with Aastra 480i's for about 14 months and at first they were fairly limited and had many issues. This all turned out to be due to the software being far before its time. As I understand it, and I may be wrong, it was due to existing customer demand that it was released arguably before its time. I came close to switching to another phone, but after the 1.3.0 firmware release, things took a serious turn. As of last week the latest firmware has been released and things are completely different. Everyone seems to agree that the 480i build and voice quality is excellent. I can not say how it compares to the Polycom or other quality phones, but with the new XML features and many other enhancements and fixes, this phone is a serious option. I now have over 40 in operation and everything is great. Since the two cheaper Aastra IP phones use near the same software, I assume they are on par. I do not know about their physical quality though. Please see the following for more information. Join the list also if you are even more interested. http://www.voip-info.org/tiki-index.php?page=Aastra+480i That said, from everything I have heard, the Polycom is one to try. I may be very happy with the 480i and likely to continue using and recommending them in the future, but I also intend on trying a Polycom in the near future and likely using them as well. Richard On 2/18/06, Michael J. Liberatore [EMAIL PROTECTED] wrote: Well the gxp-2000 has BLF, the polycom 501 does not correct?I had anastra 480i and it was prety bad, but I was going to test the 9133i for an inexpensive phone to compete with the gxp2000.The gxp2000 is notbad though, the new firmware helps a lot, but once they work out theecho bugs fully and the various minor stuff it will be a good sub $100 phone.I am yet to find a phone under $300 that's perfect... The snom360 is nice, but I have lots of problems with those too.I havent triedany polycom's though and starting to think they might be some of th ebest... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?
Basically you just plug it into an analog interface after installing the GSM chip. The voice quality is good even in my office; a sort of radio waves-black hole. Normally most cellphones just disappear when they are there.. The only problem I have so far is that the TDM400 FXO module does not seem to read the caller id. A regular phone shows it, if I switch connections. It might be a problem of configuration of the TDM card; I have looked in the wiki and googled around, but I do not know how I can change the way a zaptel card reads the callerid. I will try to upgrade to 1.2.x asap to see if this helps. Hi, Do you have any success receiving the caller id with your TDM400 FXO? I have the same problem when I connect the GSM gateway to a SPA3000 FXO line and thought this a Sipura's problem. On a phone connected to the GSM gateway I can see the callerid, but not on the Sipura's PSTN line ... Thanks, benchev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application Faxing using SIP
On 19 Feb 2006, at 06:04, Lee Howard wrote:J Poz wrote: Using an analog line is not an option for my service. My application runs on a ROOT SERVER of an ASP. So I can do anything I want to the server but I can't connect to or get external analog lines. So my options are doing faxing via the Internet (VOIP/SIP) or use a faxing service. But my experience with faxing services has not been too good as I've mentioned. Traditional faxing (not T.38) pretty much requires a lossless audio channel. Normally the best way to get this is with PSTN channels/lines through a Zap device. That said, VoIP channels can be configured such that they are also lossless. IAXmodem, for example, functions on the premise that an IAX2 channel passing over the loopback device will be lossless. I have also seen lossless SIP and IAX channels running over a WAN, but they were very specificially configured, and I wouldn't expect most connections with traditional VoIP providers to be anything near the kind of losslessness that is required for this to work well.For the most part, I suspect that those VoIP providers that promise fax support (over VoIP G.711) are doing so on a type of gamble... that ECM support of most fax machines will compensate, that they can control enough of the communication to mitigate the problem substantially, and that the remaining (say, 10%) error rate will not cause significant enough complaints from the users to cause it to be unprofitable.So, be forewarned that faxing over VoIP channels is usually not going to work extremely well for you... not unless you can mitigate the problem by creating near-lossless connections between you and the endpoint with the PSTN connection.Unless I've misunderstood the problem, your best bet is to take VOIP out of the picture, and keep your faxpurely digital up to the last possible moment. (in other words this isn't a problem for asterisk...) Can't you move the architecture about a bit ? get the ROOT server to generate a suitable PDF or TIFF of the faxthen send it over a reliable protocol (lpr/http post?) to a server that does have an analog or digital line and that is running fax software?All that said, I've had moderate success plugging an analog fax modem (in a Mac) into an ATA talking G711 to an asterisk (on the same ethernet switch) which sends it out over a PRI. This is low volume stuff. The only problemsI see are occasional retries and a total lockup when I put the Mac into sleep mode (That's an OSX bug I think).OSX does have a really nice lp-fax engine, but that is straying way off topic.T. http://www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream GXP-2000
The GXP2000 firmware is not bad for features and ease of use but still buggy. The hardware is junk to be quite honest and I don't think firmware will ever fix that. The Aastra 9133i hardware is 10x better. I have a few of both here at the moment, and I'm not sure I'd agree with that. The 9133i's handset feels much more sturdy, but the buttons on the 9133i wobble (for want of a better word) when pressed and it's difficult to determine length of travel for them. The display on the GXP2000 is significantly clearer (not just larger, the resolution seems to be better) and the buttons have firm travel limits when pressed. If only they could provide a decent weight of handset with proper sidetone, it'd be much improved. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream GXP-2000
I know it's still beta, but don't use the latest firmware in production unless you can live with an empty display after transferring a call. Only a reboot of the phone will give you text on the display again. I tested and confirmed this with 5 phones. What firmware are you running? I've just tried both attended and unattended transfer on the GXP2000 on my desk (1.0.2.8 firmware) and the display is definitely as it should be after the transfer. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 Register Problem
Hi allI have a problem to register a cisco 7960 to an asterisk 1.2.2I defined in sip.conf the next :["phonenumber"]type=friendusername="username"secret="password"host=dynamiccontext=workI am trying to catch the register requests with sip debugwith no success (empty screen).I can only catch the register messages with ngrep on host it's comming from.#U CISCO_IP:50339 - ASTERISK_IP:5060 REGISTER sip:ASTERISK_IP SIP/2.0..Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK75640688..From: sip:[EMAIL PROTECTED]: sip:[EMAIL PROTECTED]: [EMAIL PROTECTED]: 41929 REGISTER..User -Agent: CSCO/6..Contact: sip:[EMAIL PROTECTED]:5060..Content-Length: 0..Expires: 1200 #I ASTERISK_IP - CISCO_IP 3:10 E..}a.o.rC...;;.i..REGISTER sip:ASTERISK_IP SIP/2.0..Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK75640688 ..From: sip:[EMAIL PROTECTED]: sip:[EMAIL PROTECTED]: [EMAIL PROTECTED]: 41929 REGISTER..User-Agent: CSCO/6..Contact: sip:[EMAIL PROTECTED]:5060..Content-Length: 0..Expires: 1200 #U CISCO_IP:50341 - ASTERISK_IP:5060 REGISTER sip:ASTERISK_IP SIP/2.0..Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3e649d37..From: sip:[EMAIL PROTECTED]: sip:[EMAIL PROTECTED]: [EMAIL PROTECTED]: 42010 REGISTER..User -Agent: CSCO/6..Contact: sip:[EMAIL PROTECTED]:5060..Content-Length: 0..Expires: 1200 #I ASTERISK_IP - CISCO_IP 3:10 E..}a.n.rC...;;.i..REGISTER sip:ASTERISK_IP SIP/2.0..Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3e649d37 ..From: sip:[EMAIL PROTECTED]: sip:[EMAIL PROTECTED]: [EMAIL PROTECTED]: 42010 REGISTER..User-Agent: CSCO/6..Contact: sip:[EMAIL PROTECTED]:5060..Content-Length: 0..Expires: 1200 If there any way to find what's the reason why i can not register the phone ??Thanks for the help. Brings words and photos together (easily) with PhotoMail - it's free and works with Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000
On 11:07, Sun 19 Feb 06, Chris Bagnall wrote: I know it's still beta, but don't use the latest firmware in production unless you can live with an empty display after transferring a call. Only a reboot of the phone will give you text on the display again. I tested and confirmed this with 5 phones. What firmware are you running? I've just tried both attended and unattended transfer on the GXP2000 on my desk (1.0.2.8 firmware) and the display is definitely as it should be after the transfer. I tried with one phone on both * svn head, *1.2 and *1.0.9 The exact fw version for the phone is something I cannot get for you now as the phone is back to the shelve. The fw files on my http server are dated Jan 19. Maybe that's a marker. It was the latest I could get back then. It has PPPoE and BLF included. (I updated the phone to get BLF working) I'll have a look later this week if there's a new fw for the phone and start testing again. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application Faxing using SIP
On Sat, 2006-02-18 at 22:04 -0800, Lee Howard wrote: Traditional faxing (not T.38) pretty much requires a lossless audio channel. Normally the best way to get this is with PSTN channels/lines through a Zap device. That said, VoIP channels can be configured such that they are also lossless. IAXmodem, for example, functions on the premise that an IAX2 channel passing over the loopback device will be lossless. I have also seen lossless SIP and IAX channels running over a WAN, but they were very specificially configured, and I wouldn't expect most connections with traditional VoIP providers to be anything near the kind of losslessness that is required for this to work well. I have a PRI terminated in a TE110XP card on my Asterisk box. Right now we are using a separate analog line for faxing, but (for a variety of reasons) I would like to switch to sending and receiving faxes over the PRI via Asterisk. What's the recommended way to do this? The three obvious options I can think of are: 1. Connect the fax machine to an ATA and have it speak SIP or IAX to * 2. Fit a TDM400P with FXS linecard into the * box and connect the fax machine to it directly. 3. Replace the TE110XP with a multispan E1 card, connect a channel bank to the second span, and plug the fax machine into that. Option 3 can be ruled out immediately for us due to cost. Option 2 is quite appealing, but I've previously been told that running multiple Zap cards in a single machine is not a good idea. Option 1 seems like the cheapest and easiest, but I have no idea how reliably faxing will work over an ATA. Thanks for any insight. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] co-location providers in Ottawa, Canada
You can use Sprint (Group Telecom) and/or Magma. Keep us posted about the group meetings.. Thanks,Wojtek - Original Message - From: Richard OSS To: asterisk-users@lists.digium.com Sent: Sunday, February 19, 2006 12:03 AM Subject: [Asterisk-Users] co-location providers in Ottawa, Canada Anybody know ifthere are co-location providers in Ottawa, Canada? We are planning on co-locating our Asterisk conferencing server. One more thing, is there an interest in reviving the Ottawa Asterisk User Group? Seems like the original group has been inactive for quite awhile. I will volunteer to organize it. Thanks. richard ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000
On Sun, 19 Feb 2006, Michiel van Baak wrote: I tried with one phone on both * svn head, *1.2 and *1.0.9 The exact fw version for the phone is something I cannot get for you now as the phone is back to the shelve. What is the MAC address of your phones? There are hardware revisions of the gxp2000 which are known to have problems with the beta firmware. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream GXP-2000
On Sat, 18 Feb 2006, Michael J. Liberatore wrote: Well the gxp-2000 has BLF, the polycom 501 does not correct? I had an astra 480i and it was prety bad, but I was going to test the 9133i for an inexpensive phone to compete with the gxp2000. The gxp2000 is not bad though, the new firmware helps a lot, but once they work out the echo bugs fully and the various minor stuff it will be a good sub $100 phone. I am yet to find a phone under $300 that's perfect... The snom 360 is nice, but I have lots of problems with those too. I havent tried any polycom's though and starting to think they might be some of th ebest... The GXP2000 is good value for the money. It is not a great phone but for your $80 you get a lot more than one would expect. 7 programmable buttons with BLF, Backlight, dual 100bt. Stuff you dont find on some phones over twice the price... All phones have their warts, even cisco. For $80 I can live with the GXP2000's warts, grandstream do seem to be actively improving the firmware and fixing what they can. Asterisk features (mwi, blf) just work out of the box without the gyrations one has to go through for other vendors phones. I have some $200+ phones which have some serious warts and the vendors do not seem terribly interested in fixing them. Big money does not always mean good value. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HandyTone 488 ata?
On Sun, 2006-01-29 at 19:15 +, Phil Blundell wrote: Our HT386s are also a little bit prone to locking up and needing to be rebooted, but that seems to be a different problem: it occurs less often than on the HT488, and seems to be triggered by something to do with call transfers (which we never did with the 488). I've just bought an SPA-3000 to replace the HT488, though I haven't installed it yet. I'm hoping that I'll have a better experience with this one. If that works out, I might toss the 386s in favour of SPA-2000s as well. In case anybody is interested, an update on this: I replaced the HT488 and one of the '386s with an SPA-3000 and an SPA-2002 respectively, and reliability does seem to be much improved. Neither of those units have crashed yet after a couple of weeks of use, whereas the '488 would lock up almost every day and the '386 about once a week on average. Early on, I had a bit of a problem with the SPA-3000 apparently not hanging up the FXO line properly at the end of a call; this seems to have gone away after some tweaking of the line settings, but I'm still keeping it under review. I also had a bit of a battle getting the dialplan on the SPAs working right, and I suspect there might still be a couple of things wrong there. The SPAs seem to be generally more configurable than the Handytones, as well. In particular, it looks like the ring cadences are configurable, which should be a welcome relief to those of my users who complained about the American ring cadence on the Handytone. Overall, I'm happier with the SPAs than the handytones, though neither of them are entirely perfect. Oh well. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wildfire messsaging server
http://www.jivesoftware.org/ Is anyone running a wildfire messaging server on the same pc as their asterisk server? Is anyone specifically running it on an [EMAIL PROTECTED] installation? TIA, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application Faxing using SIP
Traditional faxing (not T.38) pretty much requires a lossless audio channel. Normally the best way to get this is with PSTN channels/lines through a Zap device. That said, VoIP channels can be configured such that they are also lossless. IAXmodem, for example, functions on the premise that an IAX2 channel passing over the loopback device will be lossless. I have also seen lossless SIP and IAX channels running over a WAN, but they were very specificially configured, and I wouldn't expect most connections with traditional VoIP providers to be anything near the kind of losslessness that is required for this to work well. I have a PRI terminated in a TE110XP card on my Asterisk box. Right now we are using a separate analog line for faxing, but (for a variety of reasons) I would like to switch to sending and receiving faxes over the PRI via Asterisk. What's the recommended way to do this? The three obvious options I can think of are: 1. Connect the fax machine to an ATA and have it speak SIP or IAX to * 2. Fit a TDM400P with FXS linecard into the * box and connect the fax machine to it directly. 3. Replace the TE110XP with a multispan E1 card, connect a channel bank to the second span, and plug the fax machine into that. Option 3 can be ruled out immediately for us due to cost. Option 2 is quite appealing, but I've previously been told that running multiple Zap cards in a single machine is not a good idea. Option 1 seems like the cheapest and easiest, but I have no idea how reliably faxing will work over an ATA. A rather knowledgable person suggested (off list) that faxing via the TDM400 analog card is more reliable if the fax machine is directly connected to an fxs interface module located on the same TDM400 card as the fxo module. I don't have any fxs modules to try it and have no idea whether that might be true or not, but might be worth the effort for someone that has both modules to test. If that is true, it would certainly improve the image/usability of the TDM card a bunch for small systems. (Its been a fairly major show-stopper for placing asterisk in any small business environment.) Using the TDM400 analog card with faxing via asterisk (eg, ata devices) have an extremely high failure rate (would guess better then 90% of those that have attempted it failed totally). It seems the TJ320 chip on the card gets blamed for missed/dropped packets across the pci bus, which would cause a total fax failure. To my knowledge, no one has actually proven whether that statement is true or not. The issues with all three options noted above essentially involves dropped/missed packets, regardless of whether those packets are missed sip packets across a lan/wan or via the system pci bus. Think of it as an unreliable layer-2 communications issue that you have little or no control over. If someone can send me a fxs module (for the TDM analog card), I would certainly validate the above with supporting documentation. None of the above comments apply to digital fxo interfaces such as PRI's, etc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7905 can't register
My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k
on Debian stable). It could, however, register with another
installation of Asterisk and the settings on the phone (apart from the
SIP proxy address) haven't changed since then.
On the new Asterisk box my sip.conf contains this:
[jeremy]
type=friend
regexten=801
allow=g729
host=dynamic
secret=PASSWORD
nat=yes
qualify=yes
canreinvite=no
callerid=Jeremy Malcolm 9213 0801
mailbox=1000
sip show peers shows this:
jeremy(Unspecified)D N 255.255.255.255 0UNKNOWN
sip debug shows this:
Sip read:
REGISTER sip:tardis.malcolm.id.au SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:5060
From: sip:[EMAIL PROTECTED];tag=1655991738
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
Contact: Jeremy Malcolm
sip:[EMAIL PROTECTED]:5060;transport=udp;expires=3600
User-Agent: Cisco-CP7905/1.01-030807A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.104 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.104:5060;received=220.238.97.78;rport=50048
From: sip:[EMAIL PROTECTED];tag=1655991738
To: sip:[EMAIL PROTECTED];tag=as779edf52
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 220.238.97.78:50048
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.104:5060;received=220.238.97.78;rport=50048
From: sip:[EMAIL PROTECTED];tag=1655991738
To: sip:[EMAIL PROTECTED];tag=as779edf52
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
WWW-Authenticate: Digest realm=asterisk, nonce=52e7665f
Content-Length: 0
to 220.238.97.78:50048
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Can anyone offer any advice?
TIA
--
JEREMY MALCOLM [EMAIL PROTECTED] - lawyer, IT consultant and actor.
Internet and Open Source specialist. Web site: http://www.malcolm.id.au.
host -t NAPTR 1.0.8.0.3.1.2.9.8.1.6.e164.org |awk '{print substr($7,6)}'
smime.p7s
Description: S/MIME Cryptographic Signature
___
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Re: [Asterisk-Users] Cisco 7960 Register Problem
I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : [phonenumber] type=friend username=username secret=password host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). The 7960's are one of the easiest phones to configure, so I'd have to assume you've got something very wrong in the phone's config. In the above samples you're showing quotes. If those are actually present in your config, remove them. Try something like this: [1234] type=friend username=1234 secret=mysecret host=dynamic context=work dtmfmode=rfc2833 canreinvite=no mailbox=1234 Then in your 7960 config file (macaddress.cnf), use matching entries like this: line1_name: 1234 line1_authname: 1234 line1_password: mysecret In your 7960 config file (DIPDefault.cnf), ensure the proxy address is defined as the IP adddress of asterisk: proxy1_address: 192.168.1.1 The above are the only parameters needed to make a 7960 register with asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] co-location providers in Ottawa, Canada
I think Unlimitel.ca(Embrun) offer this service. On 2/19/06, Richard OSS [EMAIL PROTECTED] wrote: Anybody know ifthere are co-location providers in Ottawa, Canada? We are planning on co-locating our Asterisk conferencing server.One more thing, is there an interest in reviving the Ottawa Asterisk User Group? Seems like the original group has been inactive for quite awhile. I will volunteer to organize it.Thanks.richard ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM GATEWAY
Hi everyone, Can anyone give me suggestions on any equipment that can connect from VOIP to a GSMgateway(channelbank that can load up to 30 sim cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's Stargate (which has a VOIP card) but the device is really a GSM-to-ISDN device. I am looking of a device that is purely VOIP to GSM Any Ideas?? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildfire messsaging server
Yes to both. It works perfectly fine - followed the instructions to the t; a few things you learn as you install :) let me know if you need any assistance or if you want us to install it for you rajeev Dean Collins wrote: http://www.jivesoftware.org/ Is anyone running a wildfire messaging server on the same pc as their asterisk server? Is anyone specifically running it on an [EMAIL PROTECTED] installation? TIA, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Test
testing first email to list ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] any doc/example for app_sms.so ?
is there any documentation or simple example around for app_sms.so to get started with it and do two simple tasks: 1. send a message to an sms-capable phone connected to an ATA 2. receive a message from an sms-capable phone and so something simple with it, even just write it to the debug screen... thanks luigi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Application Faxing using SIP
If you are sharing a box at an ASP, you might have just identified the cause of your problems. Faxing is very time sensitive. With voice, you won't notice or care if there are brief dropouts of audio. With fax, these will cause resend of the raster line (hence the long delays). If your box is shared with other apps, you may not be getting the time slices you need (very different from overall CPU power you are getting). Can you get onto your own box at the ASP? MD From: J Poz [mailto:[EMAIL PROTECTED] Sent: Saturday, February 18, 2006 11:35 PMTo: Technical Support; Asterisk Users Mailing List - Non-Commercial Discussion; Philip EdelbrockSubject: RE: [Asterisk-Users] Application Faxing using SIP MD, Using an analog line is not an option for my service. My application runs on a ROOT SERVER of an ASP. So I can do anything I want to the server but I can't connect to or get external analog lines. So my options are doing faxing via the Internet (VOIP/SIP) or use a faxing service. But my experience with faxing services has not been too good as I've mentioned. Does your company provide an affordable, reliable, and somewhat real-time faxing service? Or can you recommend one? Otherwise, I have to experiment and try to see the results I can get with doing Internet faxing. Remember my experience so far with fax service providers - single faxes take 40+ minutes to eventually be sent (and the delays are within the faxing service and and not the receiving fax line - I've researched this). Technical Support [EMAIL PROTECTED] wrote: J: We developed the mail2fax application (www.generationd.com) - so we should be able to give some insight. I think you are confusing the time to "process" the incoming (by email) fax document, and the time to fax the document. Fax over IP causes an enormous number of retries - thus delays. I would suggest you do some experimenting with an analog line connected to your asterisk box. MD From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J PozSent: Saturday, February 18, 2006 4:17 PMTo: Philip Edelbrock; Asterisk Users Mailing List - Non-Commercial DiscussionCc: J PozSubject: Re: [Asterisk-Users] Application Faxing using SIP Thanks for the information. I prefer to try to develop/configure something myself versus using an external provider. I currently use one (have tried another) and there are no guarantees on reliability, timelines, etc. I'm in need of as close to real-time response as possible (assuming the fax machineson the other end are operational). Something like within 5 minutes 85% of the time. My experience with 2 external mailtofax providers so far is terrible. Just yesterday, a simple 1 page fax took 42 minutes to finally send it. Other scenarios were 20, 26 minutes, etc. The provider tells me that they don't gaurantee anything but most of the time sending a single page fax within 5 minutes 80% of the time. I wish I would see thatbut so far I'm seeing terrible response times from the few I've tried. I also need to know status of fax (in queue, failed, etc) in real-time so my application and react appropriately (send notification to support staff, etc). I've found one that does have some sort of guarentee by the cost of through the roof and would kill my business model/plan (and the gaurantees are wishy washy). So I think I need to "control" my own destiny. And this definitely is not anything related to spam fax, etc. - legit business but right now can't fully reveal. So I'll have to research abit on IAXModem to use it. But your suggestion is a good one. Can you share what Asterisk configuration you use to both receive the iaxmodem feed and interface to the VOIP provider for such a configuration. Thanks, JPhilip Edelbrock [EMAIL PROTECTED] wrote: On Feb 18, 2006, at 11:35 AM, J Poz wrote: I have a specific business problem that I'm hoping someone has ideas and/or has already worked out a solution. My application needs to be able to automatically create and issue faxes to many different fax machines. The volume is going to be very high. And it is only about sending faxes and not receiving them. My application is hosted by an ASP but the Linux (Fedora 2) server is mine (dedicated). So the option of having PSTN lines to do faxes is not an option since I don't own nor can put anything in the data center. I found a SIP/VOIP provider that says they do faxing (and I can connect to them using my own device (meaning asterisk or something else if necessary)). Their requirement for faxing to work on their end is to make sure i send them via their voip service using G.711 codec. So I've done alot of research on faxing and asterisk and hylafax but I' m still at a loss. F or starters, what is the architecture that I need? my
RE: [Asterisk-Users] Wildfire messsaging server
Any issues with timing etc? up until now I've been very careful to run my asterisk as a standalone solution. Any issues with interfacing into other IM platforms? I'm happy to try this out but I have one client who uses MSN IM for voice a lot (I keep trying to get him to adopt skype but not interested). Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Fusion @ Gyantec Sent: Sunday, 19 February 2006 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Wildfire messsaging server Yes to both. It works perfectly fine - followed the instructions to the t; a few things you learn as you install :) let me know if you need any assistance or if you want us to install it for you rajeev Dean Collins wrote: http://www.jivesoftware.org/ Is anyone running a wildfire messaging server on the same pc as their asterisk server? Is anyone specifically running it on an [EMAIL PROTECTED] installation? TIA, Dean --- - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Application Faxing using SIP
Sorry, I didn't intend to imply I was sharing the server. It's a root server and I control everything on it. The only thing running on it is my application - it's not shared with anyone or anything else.Technical Support [EMAIL PROTECTED] wrote: If you are sharing a box at an ASP, you might have just identified the cause of your problems. Faxing is very time sensitive. With voice, you won't notice or care if there are brief dropouts of audio. With fax, these will cause resend of the raster line (hence the long delays). If your box is shared with other apps, you may not be getting the time slices you need (very different from overall CPU power you are getting).Can you get onto your own box at the ASP?MD From: J Poz [mailto:[EMAIL PROTECTED] Sent: Saturday, February 18, 2006 11:35 PMTo: Technical Support; Asterisk Users Mailing List - Non-Commercial Discussion; Philip EdelbrockSubject: RE: [Asterisk-Users] Application Faxing using SIPMD,Using an analog line is not an option for my service. My application runs on a ROOT SERVER of an ASP. So I can do anything I want to the server but I can't connect to or get external analog lines. So my options are doing faxing via the Internet (VOIP/SIP) or use a faxing service. But my experience with faxing services has not been too good as I've mentioned.Does your company provide an affordable, reliable, and somewhat real-time faxing service? Or can you recommend one? Otherwise, I have to experiment and try to see the results I can get with doing Internet faxing. Remember my experience so far with fax service providers - single faxes take 40+ minutes to eventually be sent (and the delays are within the faxing service and and not the receiving fax line - I've researched this). Technical Support [EMAIL PROTECTED] wrote: J:We developed the mail2fax application (www.generationd.com) - so we should be able to give some insight. I think you are confusing the time to "process" the incoming (by email) fax document, and the time to fax the document. Fax over IP causes an enormous number of retries - thus delays. I would suggest you do some experimenting with an analog line connected to your asterisk box.MD From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J PozSent: Saturday, February 18, 2006 4:17 PMTo: Philip Edelbrock; Asterisk Users Mailing List - Non-Commercial DiscussionCc: J PozSubject: Re: [Asterisk-Users] Application Faxing using SIP Thanks for the information. I prefer to try to develop/configure something myself versus using an external provider. I currently use one (have tried another) and there are no guarantees on reliability, timelines, etc. I'm in need of as close to real-time response as possible (assuming the fa x machineson the other end are operational). Something like within 5 minutes 85% of the time. My experience with 2 external mailtofax providers so far is terrible. Just yesterday, a simple 1 page fax took 42 minutes to finally send it. Other scenarios were 20, 26 minutes, etc. The provider tells me that they don't gaurantee anything but most of the time sending a single page fax within 5 minutes 80% of the time. I wish I would see thatbut so far I'm seeing terrible response times from the few I've tried. I also need to know status of fax (in queue, failed, etc) in real-time so my application and react appropriately (send notification to support staff, etc).I've found one that does have some sort of guarentee by the cost of through the roof and would kill my business model/plan (and the gaurantees are wishy washy). So I think I need to "control" my own destiny.And this definitely is not anything related to spam fax, etc. - legit business but right now can't fully reveal.So I'll have to research abit on IAXModem to use it. But your suggestion is a good one. Can you share what Asterisk configuration you use to both receive the iaxmodem feed and interface to the VOIP provider for such a configuration.Thanks, JPhilip Edelbrock [EMAIL PROTECTED] wrote: On Feb 18, 2006, at 11:35 AM, J Poz wrote: I have a specific business problem that I'm hoping someone has ideas and/or has already worked out a solution. My application needs to be able to automatically create and issue faxes to many different fax machines. The volume is going to be very high. And it is only about sending faxes and not receiving them. ; My application is hosted by an ASP but the Linux (Fedora 2) server is mine (dedicated). So the option of having PSTN lines to do faxes is not an option since I don't own nor can put anything in the data center. I found a SIP/VOIP provider that says they do faxing (and I can connect to them using my own device (meaning asterisk or something else if necessary)). Their requirement for faxing to work on their end is to make sure i send them via their voip service using G.711 codec. So I've done alot of research on faxing and asterisk and hylafax but I' m still
[Asterisk-Users] Intro to Asterisk VoIP telephony course - March 21st London seats still available
There are still seats open in our March 21st to 23rd Introduction to Asterisk and VoIP telephony course. More information is available at www.signate.com. Paul Mahler ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application Faxing using SIP
Phil Blundell wrote: 1. Connect the fax machine to an ATA and have it speak SIP or IAX to * This will work provided that you can create a near-lossless communication path between the ATA and the PSTN gateway (which is the Asterisk box, I assume). One way of creating that, I would expect, would be to add another ethernet card to your Asterisk server and then run a crossover cable between that interface and the ethernet interface of the ATA. You'll also need to configure the ATA to not do lots of things typically done by ATAs, like echo cancellation. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GSM GATEWAY
Why not get 30 GSM Gateway from us at £60 each and then get an asterisk or some voip gateway like A800 and then link it all up From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dumpolid Exeplish Sent: Sunday, February 19, 2006 10:54 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] GSM GATEWAY Hi everyone, Can anyone give me suggestions on any equipment that can connect from VOIP to a GSMgateway(channelbank that can load up to 30 sim cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's Stargate (which has a VOIP card) but the device is really a GSM-to-ISDN device. I am looking of a device that is purely VOIP to GSM Any Ideas?? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intro to Asterisk VoIP telephony course - March21st London seats still available
This seems pretty commercial for a non-commercial list! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Sunday, February 19, 2006 10:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Intro to Asterisk VoIP telephony course - March21st London seats still available There are still seats open in our March 21st to 23rd Introduction to Asterisk and VoIP telephony course. More information is available at www.signate.com. Paul Mahler ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application Faxing using SIP
On Sun, 2006-02-19 at 07:36 -0800, Lee Howard wrote: This will work provided that you can create a near-lossless communication path between the ATA and the PSTN gateway (which is the Asterisk box, I assume). One way of creating that, I would expect, would be to add another ethernet card to your Asterisk server and then run a crossover cable between that interface and the ethernet interface of the ATA. You'll also need to configure the ATA to not do lots of things typically done by ATAs, like echo cancellation. That's a good idea. I hadn't thought of using a crossover cable and a dedicated card like that. (Though, that said, I suspect that the datapath through our regular network switches is probably close enough to lossless for this purpose as well.) Any recommendations as to which ATAs are suitable for this purpose? I don't remember seeing a way to disable echo cancellation on either the Grandstream or the Sipura ones that I have here. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application Faxing using SIP
My 2 cents worth: I really think that a suggestion that someone in this thread gave regarding asterisk not being the right medium for this is correct. Check out http://www.tpc.int/ which implements a email2fax gateway, one you can implement for yourself, instead of providing to the public. Hylafax with some custom scripting is definitely an option. Of course, you would have to setup one pc at your own facility (home/office) where you can connect analog phone lines. Fax messages to be sent from your hosted facility can be scripted to be emailed to your email2fax gateway from there it can go on the analog network. Assumption here is that you don't mind PSTN charges for fax, which you would otherwise avoid on voip. With regards. Sanjay. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g.729 woes
Make sure you have the correct codec for your platform. If you use an optimized codec intended for another platform it would have this problem. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Firmware version 1.3.1 released for AastraIPphones
The short answer is all the officially supported configuration parameters are in the admin guide and release notes. Options that aren't documented aren't guaranteed to work between releases. So, sorry but the current documentation contains all the config options. Gareth -Original Message- From: [EMAIL PROTECTED] on behalf of Lee Archer Sent: Fri 2/17/2006 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: Firmware version 1.3.1 released for AastraIPphones Nice one it works. Is there a complete list of all the options you can use in the config files? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gareth Owen Sent: 17 February 2006 13:39 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Firmware version 1.3.1 released for Aastra IPphones The follow should work from the configuration files (aasta.cfg/MAC.cfg), although I haven't tried it... audio mode: mode Where mode is a number between 0 and 3 0 = speaker 1 = headset 2 = speaker/headset 3 = headset/speaker Gareth Lee Archer wrote: Any chance of getting a config option in that allows you set headset/speaker in the audio menu? Lee winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [slackware 10.2 and TE205P] Unknown signalling method 'pri_cpe'
hi after some testing with [EMAIL PROTECTED], i've decided to install my asterisk server on a slackware (because it's my favourite distro and it is still suggested here http://www.voip-info.org/wiki-Asterisk+Linux+Slackware) , so, i've installed the last 10.2 release, and i've recompiled the 2.6.15.4 kernel. then i've downloaded asterisk1.2.4 and zaptel1.2.4 i load module zaptel and wct2xxp (i've got a TE205P) this is my lsmod: [EMAIL PROTECTED]:/data/programmi/asterisk1.2.4/asterisk-1.2.4# lsmod Module Size Used by wct4xxp 106048 - zaptel222948 - and this are the channel in [EMAIL PROTECTED]:~# cat /proc/zaptel/1 Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 HDB3/CCS/CRC4 RED 1 TE2/0/1/1 Clear 2 TE2/0/1/2 Clear 3 TE2/0/1/3 Clear 4 TE2/0/1/4 Clear 5 TE2/0/1/5 Clear 6 TE2/0/1/6 Clear 7 TE2/0/1/7 Clear 8 TE2/0/1/8 Clear 9 TE2/0/1/9 Clear 10 TE2/0/1/10 Clear 11 TE2/0/1/11 Clear 12 TE2/0/1/12 Clear 13 TE2/0/1/13 Clear 14 TE2/0/1/14 Clear 15 TE2/0/1/15 Clear 16 TE2/0/1/16 HDLCFCS 17 TE2/0/1/17 Clear 18 TE2/0/1/18 Clear 19 TE2/0/1/19 Clear 20 TE2/0/1/20 Clear 21 TE2/0/1/21 Clear 22 TE2/0/1/22 Clear 23 TE2/0/1/23 Clear 24 TE2/0/1/24 Clear 25 TE2/0/1/25 Clear 26 TE2/0/1/26 Clear 27 TE2/0/1/27 Clear 28 TE2/0/1/28 Clear 29 TE2/0/1/29 Clear 30 TE2/0/1/30 Clear 31 TE2/0/1/31 Clear the problem is that when i strars asterisk with asterisk -c [chan_zap.so] = (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found Jan 2 00:16:57 ERROR[2059]: chan_zap.c:10606 setup_zap: Unknown signalling method 'pri_cpe' Jan 2 00:16:57 ERROR[2059]: chan_zap.c:10231 setup_zap: Signalling must be specified before any channels are. Jan 2 00:16:57 WARNING[2059]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Jan 2 00:16:57 WARNING[2059]: loader.c:554 load_modules: Loading module chan_zap.so failed! Ouch ... error while writing audio data: : Broken pipe the strange thing is that i've copied zapata.conf and zaptel.conf from the previos [EMAIL PROTECTED] installation ... [EMAIL PROTECTED]:~# cat /etc/asterisk/zapata.conf [channels] language=it context=from-pstn signalling=pri_cpe switchtype=5ess rxwink=300 callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31,32-46,48-62 [EMAIL PROTECTED]:~# cat /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 bchan = 1-15, 17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 loadzone= it defaultzone = it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp 0.0.2pre25
Hello, Is anyone successfully using spandsp 0.0.2pre25 with either asterisk 1.0.x or 1.2.4? I've built a Gentoo ebuild for this version of spandsp and app_rtxfax, and it builds, but I'm not having any luck getting it working. 99% of my test faxes fail. Reverting to 0.0.2pre20 yields a much higher success rate. I've bumped the console debugging level in logger.conf to include debug and verbose, as well as the defaults. I'm running libtiff-3.7.1, because Mr. Underwood's site recommends it, even though it's a vulnerable version of libtiff. I'm faxing from a working, production asterisk 1.0.9 + spandsp 0.0.2pre20 system to a testing asterisk 1.0.10 + spandsp 0.0.2pre25 system, and 1 of 3 things usually happens: 1.) The fax goes through (very rare in testing) 2.) The fax loops indefinitely like this: Feb 19 11:46:04 DEBUG[5089]: chan_zap.c:4059 zt_read: DTMF digit: f on Zap/1-1 Feb 19 11:46:04 DEBUG[5089]: chan_zap.c:4101 zt_read: Fax already handled Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: chan_zap.c:4059 zt_read: DTMF digit: f on Zap/1-1 Feb 19 11:46:07 DEBUG[5089]: chan_zap.c:4101 zt_read: Fax already handled Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]:
[Asterisk-Users] RE: Cisco 7905 can't register
My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k
on Debian stable). It could, however, register with another
installation of Asterisk and the settings on the phone (apart from the
SIP proxy address) haven't changed since then.
...
Can anyone offer any advice?
TIA
--
JEREMY MALCOLM [EMAIL PROTECTED] - lawyer, IT consultant and actor.
Internet and Open Source specialist. Web site: http://www.malcolm.id.au.
host -t NAPTR 1.0.8.0.3.1.2.9.8.1.6.e164.org |awk '{print substr($7,6)}'
I was having this very problem last week; perhaps you've got the same issue.
You should make sure if you're behind NAT that your SIPDefault.cnf has the 2
NAT settings enabled. I didn't figure it out right away, because I was
focusing on the 401 unauthorized message, which doesn't really indicate a
NAT problem. I can't remember what the 2 settings are offhand, but they're
in the documentation.
Michael Newton
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RE: [Asterisk-Users] MixMonitor and command
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: 17 February 2006 22:17
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MixMonitor and command
On 2/17/06, Alex Barnes [EMAIL PROTECTED] wrote:
Has anyone had any success using the MixMonitor() plus command as
nothing I have tried works.
I am using 1.2.1 I did google the archive but couldn't see any
mention
of anyone using this. What I am hoping to do is run a macro on
hangup,
current method I am using seems to miss some calls 5% of calls fail
to
mix / convert to mp3 etc. Was hoping that MixMonitor would fix
this.
exten = s,n,MixMonitor(${CALLDIR}${CALLFILENAME}.wav||System(touch
/tmp/test${UNIQUEID}))
exten = s,n,Answer
exten = s,n,SayDigits(1234)
exten = s,n,StopMonitor()
exten = s,n,Hangup()
Output:
-- Executing MixMonitor(Zap/1-1,
/tmp/callrec/20060217-212722-1-IN.wav||System(touch
/tmp/test1140211642.11373)) in new stack
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing SayDigits(Zap/1-1, 1234) in new stack
-- Playing 'digits/1' (language 'en')
== Begin MixMonitor Recording Zap/1-1
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/3' (language 'en')
-- Playing 'digits/4' (language 'en')
-- Executing StopMonitor(Zap/1-1, ) in new stack
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (macro-test-script, s, 14) exited non-zero on
'Zap/1-1' in macro 'test-script'
== Spawn extension (from-outside-547551-tl-allhours, s, 1) exited
non-zero on 'Zap/1-1'
== End MixMonitor Recording Zap/1-1
== Executing [System(touch /tmp/test1140211642.11373)]
-- Hungup 'Zap/1-1'
However listing /tmp reveals no files. Running macros that only
print
NoOp's don't work either.
Alex -
The command is a system command already that is spawned by
MixMonitor. It's not the command you would expect to run from within a
dial plan itself.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
___
Thanks very much works like a charm now.
I will add this to the WIKI.
Cheers
Alex
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RE: [Asterisk-Users] GSM GATEWAY
Because there are cheaper solutions than purchasing 30 gateways that have an RJ11. S/He (sorry abd with names) would then have to get a channel banker. This is a lot more costly than some solutions out there. --- Sam Tam [EMAIL PROTECTED] wrote: Why not get 30 GSM Gateway from us at £60 each and then get an asterisk or some voip gateway like A800 and then link it all up _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dumpolid Exeplish Sent: Sunday, February 19, 2006 10:54 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] GSM GATEWAY Hi everyone, Can anyone give me suggestions on any equipment that can connect from VOIP to a GSM gateway (channelbank that can load up to 30 sim cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's Stargate (which has a VOIP card) but the device is really a GSM-to-ISDN device. I am looking of a device that is purely VOIP to GSM Any Ideas?? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000
My GXP-2000 is currently collecting dust. I had several issues with it. Mainly echo while on speaker. The other person can barely mae out what you are saying. Another issue was if the phone recieved to many calls it would just freeze up and I had to pull out the plug. Again I have not used it in a while. There may have been firmware updates since. Just my $0.02. Dovid --- Mimmus [EMAIL PROTECTED] wrote: Hi, I'm going to propose to my boss the buying 15 Grandstream GXP-2000 phones. - Is it a good choice (budget limit of 100 Euro/phone is mandatory)? - Can be a profitable business the direct buying of 50 phones (to save other money) or is it a risk? Thanks in advance -- Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream GXP-2000
I have both as well, I mostly agree with you about the display. The buttons are ok but not great on either. At the end of the day a phone is for talking and listening and the 9133i is far superior in that regard. Both the handset and speaker phone on the 9133i are the same as the 480i and are far superior to the GXP2000 IMHO. -Original Message- From: Chris Bagnall [mailto:[EMAIL PROTECTED] Sent: Sunday, February 19, 2006 3:04 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Grandstream GXP-2000 The GXP2000 firmware is not bad for features and ease of use but still buggy. The hardware is junk to be quite honest and I don't think firmware will ever fix that. The Aastra 9133i hardware is 10x better. I have a few of both here at the moment, and I'm not sure I'd agree with that. The 9133i's handset feels much more sturdy, but the buttons on the 9133i wobble (for want of a better word) when pressed and it's difficult to determine length of travel for them. The display on the GXP2000 is significantly clearer (not just larger, the resolution seems to be better) and the buttons have firm travel limits when pressed. If only they could provide a decent weight of handset with proper sidetone, it'd be much improved. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g.729 woes
Some people have to stap on others to make them selves feel good. Very unfortunate. --- Rusty Dekema [EMAIL PROTECTED] wrote: I don't think it takes a great leap of the imagination to infer that Mr. Kennedy is in fact having the problem he describes and that, although it may not be 100% standard and correct usage, the question mark at the end of his sentence is intended to ask why this problem might be happening. If you want to criticize his English grammar and writing skills, why not come out and say so? If you want him to provide more specific information, why not say so and tell him what information you want? Sure, he should have provided more details on his situation, but is it really necessary to take snide potshots at people on these lists? -Rusty On 2/17/06, Matt [EMAIL PROTECTED] wrote: Steve, Sorry but only you would know if you have Digium licenses and if when making a call it's only heard in one direction. I can not tell you if you are infact having this problem. What problem are you having? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] salesforce
Interesting discovery on the salesforce appexchange https://www.salesforce.com/appexchange/detail_overview.jsp?NavCode__c=MF3fid=a033000NIcAAAW FEATURES VoIP w/integrated PBX/ACD/Contact Center: Call routing, with screen pop, that distributes sales/support phone and IM inquiries to agents. Desktop and Browser Share: Presentation and collaboration tools that allow you to share, push, and whiteboard with co-workers and customers. Reporting and Analytics: Real-time call monitoring, voice and IM recording. Real time call detail and billing information. RESOURCES Worksmart Presentation Worksmart Data Sheet Worksmart Customization Guide PRICING Worksmart is available in Pro, Office or Enterprise versions to salesforce.com subscribers starting at $20 per user/month Please contact us at [EMAIL PROTECTED] or call 800.805.0558 x1 for more information Pricing is also available at http://www.worksmartcentral.com DESCRIPTION Pandora Networks Worksmart eliminates the need for non-compatible and point telephony, messaging and collaboration solutions. Our all-in-one communications service includes an integrated PBX, ACD, VoIP, video, messaging and collaboration services that are now enabling real time communication from within the Salesforce.com service. The desktop application integrates hunt groups, presence, soft phone, private IM, and even secure access to AOL, Yahoo, and MSN IM networks. The Web Dashboard provides account, user and phone number provisioning, real-time billing, account management, and end user control. Subscribers can now login to their Salesforce.com and Worksmart accounts via single sign providing one source for CRM data AND access to communication services and records such as call detail records, IM conversation archives and voicemail. Inbound calls trigger a screen pop or caller ID that presents the corresponding contact and account information while outbound calls can be made via click to dial from within a contact or account record. In addition, IM conversations are automatically archived and all inbound and outbound calls are time stamped and logged. Looks like a great customization of an existing platform onto voip. Anyone actually heard of Pandora Networks before? Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HandyTone 488 ata?
On Feb 19, 2006, at 6:06 AM, Phil Blundell wrote: Overall, I'm happier with the SPAs than the handytones, though neither of them are entirely perfect. Oh well. Thanks for the update... I am being told by the freaks at Grandstream that there will be a firmware update forthcoming to try to resolve the issues with the HT-488. Promises, Promises. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g.729 woes
On Feb 19, 2006, at 9:41 AM, Dovid Bender wrote: Some people have to stap on others to make them selves feel good. Very unfortunate. Some people have no sense of humor. Very unfortunate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [slackware 10.2 and TE205P] Unknown signalling method 'pri_cpe'
nik600 wrote: after some testing with [EMAIL PROTECTED], i've decided to install my asterisk server on a slackware (because it's my favourite distro and it is still suggested here http://www.voip-info.org/wiki-Asterisk+Linux+Slackware) , so, i've installed the last 10.2 release, and i've recompiled the 2.6.15.4 kernel. then i've downloaded asterisk1.2.4 and zaptel1.2.4 [chan_zap.so] = (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found Jan 2 00:16:57 ERROR[2059]: chan_zap.c:10606 setup_zap: Unknown signalling method 'pri_cpe' That's your problem. You don't have libpri installed. Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildfire messsaging server
On an Asterisk server- yes. [EMAIL PROTECTED] - not me. PaulH - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, February 20, 2006 1:17 AM Subject: [Asterisk-Users] Wildfire messsaging server http://www.jivesoftware.org/ Is anyone running a wildfire messaging server on the same pc as their asterisk server? Is anyone specifically running it on an [EMAIL PROTECTED] installation? TIA, Dean ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ARI 0.06
You are wonderful !!! for this bug, I noticed later on that by removing the second path in the monitor folder ... I didn't get any error ... the script was searching inside a file, thinking that it could be a directory where recordings were. Anyway, Again, Thanks a lot, JM On 2/17/06, Dan Littlejohn [EMAIL PROTECTED] wrote: On 2/17/06, Jean-Marc Salsa [EMAIL PROTECTED] wrote: Hi ! I always use your ARI through AAH, and indeed nice job ! A few comment : - I have seen that we could use ARI only for the Call Monitor by setting a value. would it be possible to do the same for only Voicemail ... indeed, we are using Asterisk only for Voicemail, and this would be so good only to present this tab to people ... ( And in Settings page also, hiding the Call Monitor Settings part here too ) - Same for help ( to show it or not ) I have installed it on our AAH 1.3 version and here are the error messages I get : Call Monitor Page (Only the first message on each page shows the Play link): Warning: is_dir(): Stat failed for /var/lib/asterisk/bin/archive_recordings/ (errno=20 - Not a directory) in /var/www/html/recordings/includes/bootstrap.inc on line 113 Settings Page (Didn't try to apply new settings): Warning: Invalid argument supplied for foreach() in /var/www/html/recordings/modules/settings.module on line 434 Warning: Invalid argument supplied for foreach() in /var/www/html/recordings/modules/settings.module on line 473 Warning: Invalid argument supplied for foreach() in /var/www/html/recordings/modules/settings.module on line 577 I hope you won't take these comments as critics, you are really doing a GREAT job ! Asterisk was really lacking this application part ! Thanks again, And all the best ! Jean-Marc On 2/17/06, Dan Littlejohn [EMAIL PROTECTED] wrote: ARI(Asterisk Recording Interface) has reached another milestone. The project is starting to become a full featured user portal and handle all the common errors that people seem to have.This release supports: call monitor page – new features include column sorting and filter small duration calls in addition to the ability to listen to call monitor recordings voicemail page – allows voicemail message listening and management handset feature code help page - I can never remember them all user settings web interface - that allows setting call fowarding, voicemail email and pager, voicemail password, and call monitor recording There are also alot of i18n translations now, although with all the rework of the code many are now somewhat broken and need to be updated.If you speak one of the following, email and I will send you the page to translate or updating to the appropriate ari.po page and returning it to me would be very helpful. German Greek Spanish French Hebrew Hungarian Italian Portuguese Swedish If you would like to translate ARI into another language, I would be happy to support it. Loaded into AMP CVS and also here: www.littlejohnconsulting.com?q=ari If you have a chance, take a look.Comments and suggestions are welcome. Dan 512.791.0137 www.littlejohnconsulting.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jean-Marc:Thanks for the feedback.I have addressed these issues they areavailable on my website and have been checked into AMP cvs. I have added a setting to the /recording/includes/main.conf file.$ARI_DISABLED_MODULES = ; allows forindividual modules to be disabled (they are truemodules though, and you can just delete them from the /recordings/modules directory)the is_dir error is a PHP bug. http://groups.google.com/group/mailing.www.php-dev/browse_frm/thread/1b5b94e775b70cdb/877e4406600a8121?lnk=stq=Warning%3A+is_dir()%3A+Stat+failed+for+errno%3D20+-+Not+a+directoryrnum=1hl=en#877e4406600a8121 But, I think I was able to suppress the error.The settings page errors have been corrected.Thanks;Dan512.791.0137www.littlejohnconsulting.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream GXP-2000
So lets pool our knowledge so next time we all get a perfect phone :) Phones I have used: GXP2000: We all know about this one, lots of features but you get what you pay for Echo, hums, old hardware revisions have lots of problems (screen, etc). The upside includes lots of features, BLF, 4 account support, 100Mb switch, firmware is worked on often. Linksys 941: Overall a great phone, stable solid firmware, heavy built, awesome light up dual color buttons, good sound quality. Cons: 1 switch port (new model has 2), you have to pay extra for 4 account support, no firmware upgrades although it could be because it works very well as is, no blf/speed dial buttons at all which makes it better for a call center. Snom 360: My favorite phone very well built, new firmwares all the time, xml support, overall a stable phone but still has its problems. Upside is nice screen, awesome blue light up, 12 BLF buttons, all the buttons on the phone can be reprogrammed, 2 port switch, heavy built handset, excellent sound quality, expandable. Cons: firmware isnt perfect by a long shot, can completely freeze, doesn't like asterisk's sip rules, some phones have a hum problem, price. UT Starcom F1000 Wifi: Nice little phone, customers love the way it looks, sound quality sucks, firmware sucks, range sucks, battery life is great, it needs work but with better firmware it could be a descent sub $150 wifi phone. Astra 480i CT: I bought this phone cause I liked the idea of an in expensive cordless that came with it, when I got it the 480i screen was shot, it was all dark and could only be used for minutes, so I didn't get much use out if it, the cordless didn't have its own sip registrations, the lines were linked to the base, and since I wanted the cordless to be called directly I decided to get rid of it. So that's my input, any other input would be helpful for all. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Sunday, February 19, 2006 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream GXP-2000 My GXP-2000 is currently collecting dust. I had several issues with it. Mainly echo while on speaker. The other person can barely mae out what you are saying. Another issue was if the phone recieved to many calls it would just freeze up and I had to pull out the plug. Again I have not used it in a while. There may have been firmware updates since. Just my $0.02. Dovid --- Mimmus [EMAIL PROTECTED] wrote: Hi, I'm going to propose to my boss the buying 15 Grandstream GXP-2000 phones. - Is it a good choice (budget limit of 100 Euro/phone is mandatory)? - Can be a profitable business the direct buying of 50 phones (to save other money) or is it a risk? Thanks in advance -- Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000
We had to stop offering the GXP-2000 due to all the same issues mentioned above. Really not for business use. Have had good results with Linksys SPA-941.On 2/19/06, mustardman29 [EMAIL PROTECTED] wrote: I have both as well,I mostly agree with you about the display.The buttons are ok but not greaton either.At the end of the day a phone is for talking and listening and the 9133i isfar superior in that regard.Both the handset and speaker phone on the 9133i are the same as the 480i and are far superior to the GXP2000 IMHO. -Original Message- From: Chris Bagnall [mailto:[EMAIL PROTECTED]] Sent: Sunday, February 19, 2006 3:04 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Grandstream GXP-2000 The GXP2000 firmware is not bad for features and ease of use but still buggy.The hardware is junk to be quite honest and I don't think firmware will ever fix that.The Aastra 9133i hardware is 10x better. I have a few of both here at the moment, and I'm not sure I'd agree with that. The 9133i's handset feels much more sturdy, but the buttons on the 9133i wobble (for want of a better word) when pressed and it's difficult to determine length of travel for them. The display on the GXP2000 is significantly clearer (not just larger, the resolution seems to be better) and the buttons have firm travel limits when pressed. If only they could provide a decent weight of handset with proper sidetone, it'd be much improved. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fwd: Which ATA device do you recommend?
Are the PAP2's you can get branded vonage at staples for free after rebate still hackable? I read that you cant do it beyond a certain firmware but wasn't sure if it had to be connected to the internet for that download or if it ships with that now -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Greenberg Sent: Saturday, February 18, 2006 4:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fwd: Which ATA device do you recommend? For single and two-port applications, I've had very good luck with Sipura 2000s. Now available as Linksys PAP2-NA. /edg --On Wednesday, February 15, 2006 3:08 PM + Marco Mouta [EMAIL PROTECTED] wrote: -- Forwarded message -- From: Marco Mouta [EMAIL PROTECTED] Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: [EMAIL PROTECTED] Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me their advice based on their experiencies. Thanks to all, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: SPA-941 stutter tone
Stutter tone has been used for years, you can dial whenever you want -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Meredith Sent: Friday, February 17, 2006 3:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: SPA-941 stutter tone Jock W. Shirey [EMAIL PROTECTED] wrote: I just double checked my SPA-841. You can change the dial tone in the Web config on the Regional page. I just copied the Dial Tone: to the MWI Dial Tone field and it didnt stutter after that. I'm not sure if its the same with the 941, but i've heard the phone configs are similar. Hey, I never thought of that. One thing to check: I always assumed (but never checked) that you couldn't dial until the stutter stopped, and it gave you the normal dial tone. Is this true? If so, it will be very confusing when you try to dial when you have voice mail. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA-941 hint
Where would it display the status? There are no BLF buttons... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matteo Piazza Sent: Friday, February 17, 2006 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SPA-941 hint Hi Have someome a solution to use the hint function to have the signalling of the status of a extension on the SPA-941 phone ? Matteo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] one way / irratic voice over iax and g729
So you have 2 asterisk systems connected, I am doing this for the first time. Any tips you can give me besides whats on the wiki? I am not sure the best way to set it up, I want to be able to have the 2 locations act as 1 over their internet connection to each other, I was planning to use vpn... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Dinnerville Sent: Friday, February 17, 2006 4:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] one way / irratic voice over iax and g729 Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP) Asterisk1 -(IAX) Asterisk2 -(SIP) Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there is one way voice whereby the B party cannot hear the A party, however the A party can hear the B party fine. Sometimes there is no audio for the B party, other times the B party can hear the A party but it is very broken up and stuttery, with only parts of the words coming through. The calls also work fine when using g711 from the A party. Asterisk2 is running a couple of TDM04B's so there is a physical timing device on that side and Asterisk1 is running ztdummy on a 2.6 kernel - so there is timing on that side also (??) Have done a fair bit of searching on this one, and as it only happens with g729 (both systems have the licensed codecs installed) it is a bit of a head scratcher - has anyone else experiencved this? Or does anyone have any feedback? Cheers, Ben ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line Dropouts on E405P
Hi,
We have a Ericsson BP250 Phone system setup witht
he following configuration
Telco - Asterisk E405P -
BP250
The system seem to work perfectly on 1.0.9 for a
very long time but there is some functionality we wanted to take advantage of in
the 1.2 version branch so we upgraded.
Currently running
Asterisk 1.2.4
Zaptel 1.2.3 (noticed 1.2.4 is out will upgrade
next weekend)
Libpri 1.2.2
The problem we are getting is wierd but
:-
Sorry about the timings looking wierd but you have
to allow a fudge factor of anywhere upto 12 hours when dealing with reports from
on-site personel.
* Wednesday ~ 9.30am All calls drops for 1 second,
then back online, then ~5 minutes later, same thing* Thursday ~
10.30am All calls drops for 1 second, then back online, then ~5 minutes later,
same thing
* Friday ~ 11.30am All calls drops for 1 second,
then back online, then ~5 minutes later, same thing
Just before it drops out the calls sound a little fuzzy.
There is no warning messages on console.Error log (which seem to
correspond to drops outs):Feb 17 11:30:08 WARNING[2566] chan_zap.c: No
D-channels available! UsingPrimary channel 47 as D-channel
anyway!Feb 17 11:36:12 WARNING[2565] chan_zap.c: No D-channels
available! UsingPrimary channel 16 as D-channel anyway!
D-Channel 47 relates to thesocket which is connected to the BP250,
D-Channel 16 relates to the socket connected to the telco.
I really don't want to have to drop back to 1.0.9 if i can avoid it.
Log files and settings :-
Logger.conffull =
notice,warning,errorZaptel.confspan=1,1,0,ccs,hdb3,crc4bchan=1-15dchan=16bchan=17-31span=2,0,0,ccs,hdb3,crc4bchan=32-46dchan=47bchan=48-62span=3,0,0,ccs,hdb3,crc4bchan=63-77dchan=78bchan=79-93span=4,0,0,ccs,hdb3,crc4bchan=94-108dchan=109bchan=110-124
Zapata.conf
[channels]context=defaultmusiconhold=defaultswitchtype=euroisdnusecallerid=yescidsignalling=v23cidstart=polarityhidecallerid=nocallwaiting=nousecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesechotraining=800rxgain=0.0txgain=0.0
group=1context=te405p-intelstrapridialplan=localsignalling=pri_cpe;overlapdial=yescallerid=asreceivedchannel=1-15,
17-31group=4context=te405p-frombp250pridialplan=localsignalling=pri_netoverlapdial=yescallerid=asreceivedchannel=32-46,
48-62
Extensions.conf (Sorry for it being so large, most of the rest it of is in
other files)
[default]exten = s,1,Dial(SIP/5552,45,t)
[dialstring]
exten = i,1,Playback(invalid)exten = i,2,Hangupexten =
t,1,Hangup
[atp-out]
exten = _8X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]);exten =
_8X.,1,dial(SIP/${EXTEN:[EMAIL PROTECTED],30)exten =
_8X.,2,Congestion
exten =
_9X.,1,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN:1})exten
= _9X.,2,Congestionexten = _9X.,3,Hangup
[from-callpacket]
exten = 17025541498,1,Answerexten =
17025541498,2,Dial(SIP/557)exten = 17025541498,3,Hangup
[atp-in]
exten = 30182849,1,SetMusicOnHold(record)exten =
30182849,2,Dial(SIP/551,45,t)exten = 30182849,3,Voicemail,u551exten
= 30182849,103,Voicemail,b551
exten = s,1,Dial(SIP/3332,45,t)
[te405p-frombp250]
exten =
_321X.,1,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN:3})
include = to-sipinclude = parkedcallsinclude =
record-transferinclude = atp-outinclude = voicerecinclude
= lm1_functionsinclude = te405p-outtelstra
[te405p-tobp250]
#include extensions_te405p-tobp250.conf
[te405p-intelstra]
#include extensions_te405p-intelstra.confinclude = to-sip
[te405p-outtelstra]
#include extensions_te405p-outtelstra.confinclude =
dialstring
include = js_play_ael
[from-sip]exten =
555,1,dial(SIP/username:[EMAIL PROTECTED]/0732822922)
exten = 881,1,Dial(Zap/G4/38165912)exten =
982,1,Dial(Zap/G4/38166400)exten = 983,1,Dial(Zap/G4/38105000)exten
= 984,1,Dial(Zap/G4/5483)exten =
985,1,Dial(Zap/G4/5912)exten = 986,1,Dial(Zap/G4/5760)exten
= 987,1,Dial(Zap/G4/5765)exten =
988,1,Dial(Zap/G4/1006)exten = 989,1,Dial(Zap/G4/5947)exten
= 55,1,Dial(Zap/G1/0423813901)
exten = s,1,Dial(SIP/3332,45,t)
include = atp-outinclude = lm1_functionsinclude =
from-callpacketinclude = to-sipinclude =
te405p-tobp250include = te405p-outtelstrainclude =
record-transferinclude = parkedcallsinclude = voicerec
[record-transfer]
exten = _32XX,1,SetVar(DDATE=${TIMESTAMP})exten =
_32XX,2,SetVar(CALLFILENAME=/mnt/asterisk/pub/newbiz/${DDATE:0:8}/${EXTEN:1}/${EXTEN:1}-${TIMESTAMP})exten
= _32XX,3,Monitor(gsm,${CALLFILENAME},m)exten =
_32XX,4,Dial(ZAP/g4/${EXTEN:1})exten = _32XX,5,Congestionexten
= _32XX,105,Congestion
exten =
_34XX,1,SetVar(CALLFILENAME=/mnt/asterisk/5xxx/CallTo-${EXTEN:1}-${TIMESTAMP})exten
= _34XX,2,Monitor(gsm,${CALLFILENAME},m)exten =
_34XX,3,Dial(ZAP/g4/${EXTEN:1})exten = _34XX,4,Congestionexten
= _34XX,104,Congestion
exten =
_399X.,1,Dial(IAX2/username:[EMAIL PROTECTED]/0011${EXTEN:3})exten
= _399X.,2,Congestionexten = _399X.,3,Hangup
[voicerec]
exten = 381,1,Festival('Please record your
RE: [Asterisk-Users] Line Dropouts on E405P
I did some testing - More
Information - Hope this helps...
Next thing to try is to maybe move the port
that the Asterisk - BP250 (Group 1/D-Channel 16) resides on and see if that
makes a difference.
If I call 30 numbers from Asterisk --
BP250
only 28 connect and get the following
in the log file:Feb 19 08:46:22 NOTICE[13902] app_dial.c: Unable to
create channel of type'Zap' (cause 34 - Circuit/channel congestion)Feb
19 08:46:22 NOTICE[13902] app_dial.c: Unable to create channel of type'Zap'
(cause 34 - Circuit/channel
congestion)
If I call 15 extensions
via
Asterisk - Telstra - Asterisk
- BP250
I get the following in the log
file:Feb 19 08:51:41 WARNING[2565] chan_zap.c: Ring requested on channel
0/15already in use on span 1. Hanging up owner.Feb 19 08:51:42
WARNING[2565] chan_zap.c: Ring requested on channel 0/14already in use on
span 1. Hanging up owner.Feb 19 08:51:42 WARNING[2565] chan_zap.c:
Ring requested on channel 0/13already in use on span 1. Hanging up
owner.Feb 19 08:51:42 WARNING[2565] chan_zap.c: Got restart ack on channel
0/8span 1 with owner
As I dial the 15 as above I get the
following in the CLI:asterisk1*CLI -- Executing
Dial("SIP/3332-c760","Zap/g1/38165901Zap/g1/38165902Zap/g1/38165903Zap/g1/38165904Zap/g1/38165905Zap/g1/38165906Zap/g1/38165907Zap/g1/38165908Zap/g1/38165909Zap/g1/38165910Zap/g1/38165911Zap/g1/38165912Zap/g1/38165913Zap/g1/38165914Zap/g1/38165915")
in new stack -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165901 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165902 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165903 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165904 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165905 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165906 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165907 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165908 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165909 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165910 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165911 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165912 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165913 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165914 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165915!! Got reject for frame
77, retransmitting frame 77 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 78 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 79 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 80 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 81 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 82 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 83 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 84 now, updating n_r! -- Zap/3-1 is
proceeding passing it to SIP/3332-c760 -- Zap/2-1 is
proceeding passing it to SIP/3332-c760 -- Zap/1-1 is
proceeding passing it to SIP/3332-c760 -- Zap/7-1 is
proceeding passing it to SIP/3332-c760 -- Zap/6-1 is
proceeding passing it to SIP/3332-c760 -- Zap/5-1 is
proceeding passing it to SIP/3332-c760 -- Zap/4-1 is
proceeding passing it to SIP/3332-c760 -- Channel 0/12,
span 1 got hangup -- Forcing restart of channel 0/12 on
span 1 since channel reported inuse -- Zap/11-1 is
proceeding passing it to SIP/3332-c760 -- Zap/10-1 is
proceeding passing it to SIP/3332-c760 -- Hungup
'Zap/12-1'!! Got reject for frame 86, retransmitting frame 86 now, updating
n_r!!! Got reject for frame 86, retransmitting frame 87 now, updating
n_r!!! Got reject for frame 86, retransmitting frame 88 now, updating
n_r! -- Zap/9-1 is proceeding passing it to
SIP/3332-c760 -- Accepting call from '738166400' to
'38165903' on channel 0/19, span 1 -- Executing
SetMusicOnHold("Zap/19-1", "record") in new stack --
Zap/8-1 is proceeding passing it to SIP/3332-c760 --
Executing Dial("Zap/19-1", "Zap/g4/38165903|6000|t") in new
stack -- Requested transfer capability: 0x00 -
SPEECH -- Called g4/38165903 --
Accepting call from '738166400' to '38165902' on channel 0/22, span
1 -- Executing SetMusicOnHold("Zap/22-1", "record") in new
stack -- Executing Dial("Zap/22-1",
"Zap/g4/38165902|6000|t") in new stack -- Requested
transfer capability: 0x00 - SPEECH -- Called
g4/38165902 -- Zap/32-1 is proceeding passing it to
Zap/19-1 -- Accepting call from '738166400' to '38165901'
on channel 0/17, span 1 -- Executing
SetMusicOnHold("Zap/17-1", "record") in new stack --
Executing Dial("Zap/17-1", "Zap/g4/38165901|6000|t") in new
stack
RE: [Asterisk-Users] Asterisk and Snom 360
Ok here it is, just remember who hooked you up :) But I don't see anything about fixing a crashing problem that you described in 5.3 I am running this on 1 phone and 5.3 on 3 others, the ones with 5.3 seem perfect, the one with 5.3.3 actually locked up once doing a transfer. Release 5.3.3: o GUI: fixed DND o GUI: fixed bug in displaying old voice mail messages o SIP: display local LED status for shared lines o WEB: + in settings value isn't anymore replaced by its hex value on settings dump web interface page o WEB: further enhanced french translation o SRTP: fixed bug with auto-answer o GUI: setting_server can be set manually via GUI menu (snom360) o GUI: ringer device should not switch to speaker if headset is enabled o GUI: dkeys (e.g. Redial, Retrieve) are working in edit number state, too o SETTINGS: if setting_server is IP:port only, make a valid URL out of it o SIP: display local LED status for shared lines o GUI: Shared Lines can be mapped to LEDs o LID: random number generated from random audio data http://fox.snom.com/download/snom360-5.3.3b-SIP-j.bin -Mike Mike240se -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Krief Sent: Friday, February 17, 2006 1:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Snom 360 Indeed - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 15, 2006 11:37 PM Subject: Re: [Asterisk-Users] Asterisk and Snom 360 On Wed, 15 Feb 2006, Olivier Krief wrote: Do not use 5.3 but 5.3.3 instead as major crashes occur with 5.3. http://www.snom.com/firmware.html#1641 5.3.3 is not available for public download... -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's
I had voicepulse connect but had to transfer IAX2 had non stop drop outs in audio all the time. Tried everything to fix it, even with 14ms ping times it just didnt want to work right. I never figured out why, just canceled. Although i didnt like the no-name on incoming caller id either though, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of andrew matthewsSent: Tuesday, February 14, 2006 8:52 PMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's http://connect.voicepulse.netThey support astrisk, with iax2 :) On 2/14/06, Jim Robinson [EMAIL PROTECTED] wrote: Hi Folks,Can anyone give me some good recommendations for VoIP providrs thatsupport Asterisk PBX's?We're based in Georgia and I having a hard timefinding anyoneRegards,JimPS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I install speex for asterisk?
Do you have allow=speex in your codecs list in either sip.conf or iax.conf? if not this this could be the reason. Also, Speex won't get selected if its not the primary codec on either side's call initiation. In other words you allow list should look like this disallow=all allow=speex allow=blah allow=blah When you make a SIP call you will be able to force the other side into speex if they suport it. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Jesus E Zepeda wrote: Elaborating a little more I checked for files suggested by Matthew Roth: If the build goes as planned, the /codecs directory will contain three speex-related files: - codec_speex.c - codec_speex.o - codec_speex.so Then ran the show modules command and now codec_speex shows as loaded by asterisk! But still cannot make or receive calls using speex. I am investigating with my VOIP provider.. Thanks to all of you. -Original Message- From: Jesus E Zepeda [mailto:[EMAIL PROTECTED] Sent: Friday, February 17, 2006 09:54 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How do I install speex for asterisk? Mark: I did so, but that did not make asterisk to integrate speex. Do I have to tweak something in speex after installation? This is some of asterisk output when I try to use speex: -- Accepting AUTHENTICATED call from 192.168.2.32: requested format = speex, requested prefs = (), actual format = speex, host prefs = (speex|ilbc|gsm), priority = mine -- Executing Macro(IAX2/ext2-2, outbound|14802012944) in new stack -- Call accepted by 66.234.228.160 (format speex) -- Format for call is speex -- IAX2/66.234.228.160:4569-5 is circuit-busy -- Hungup 'IAX2/66.234.228.160:4569-5' Feb 17 09:20:42 WARNING[1811]: chan_iax2.c:1717 attempt_transmit: Max retries exceeded to host 66.234.228.166 on IAX2/66.234.228.166:4569-9 (type = 6, subclass= 1, ts=8, seqno=0) -- Hungup 'IAX2/66.234.228.166:4569-9' == No one is available to answer at this time (1:0/0/0) Feb 17 09:20:52 WARNING[2508]: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context 'internal' -- Hungup 'IAX2/ext2-2' -- Registered IAX2 'ext1' (AUTHENTICATED) at 192.168.2.31:4569 Feb 17 09:30:42 NOTICE[1811]: chan_iax2.c:5673 update_registry: Restricting registration for peer 'ext1' to 60 seconds (requested 300) -Original Message- From: Mark Phillips [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 17:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How do I install speex for asterisk? If you did a make install with speex then everythings where it should be. Just do a make; make clean with asterisk and all will be fine. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Jesus E Zepeda wrote: Huuu! I never expected you had to recompile asterisk to add a codec. But if that is what it takes, we'll do it. I noticed that asterisk makes reference to some speex.c in the makefile file. In some of those references I saw the actual speex.c file in the paths specified. A couple of them missing by the way. That could be why speex was never taken by asterisk. Mike, does speex have to be copied to a specific directory, then compiled and installed before re-compiling and re-installing asterisk? I appreciate you took your time to reply. Regards, Jesus -Original Message- From: Mike Pollitt [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 15:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How do I install speex for asterisk? You need to recompile Asterisk itself after installing Speex. Do a make clean, make, make install. I usually stop asterisk before that last step, by the way! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus E Zepeda Sent: Friday, 17 February 2006 5:58 AM To: Asterisk User List Subject: [Asterisk-Users] How do I install speex for asterisk? Hi, everybody: I enabled speex in my asterisk box (iax.conf), but no phone call went throug. At the asterisk console, I used the show modules command and it did not show the speex codec in the list. So, I downloaded the speex codec from speex.org, v1.0.5, compiled and installed in my asterisk machine. What I still don't know is: what do I need to do from the asterisk side to make it available? I just downloaded it to a directory, compiled and installed thinking that by doing a restart to asterisk it would some how know where to load it from. Any hints are appreciated Regards, Jesus E. Zepeda ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Snom 360
Still beta, but we could not make it crash any more...: We would be happy about the feedback from volunteers:-) http://fox.snom.com/download/snom320-5.3.6a-beta-SIP-j.bin http://fox.snom.com/download/snom320-5.3.6b-beta-SIP-j.bin http://fox.snom.com/download/snom360-5.3.6a-beta-SIP-j.bin http://fox.snom.com/download/snom360-5.3.6b-beta-SIP-j.bin Release 5.3.6: o LID: made sure audio channels are off in idle mode under all scenarios Release 5.3.5: o GUI: added cwi ringer indication o GUI: fixed unnecessary dialog state switches on shared line offhook o GUI: status led for missed calls o SIP: RAck in PRACK was buggy o SIP: added call pickup for shared lines Release 5.3.4: o SIP: added +sip.rendering parameter for BLA hold/resume NOTIFYs o SIP: NOTIFYs with subscription-state: terminated remove the subscription ~~~ Christian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Sunday, February 19, 2006 6:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk and Snom 360 Ok here it is, just remember who hooked you up :) But I don't see anything about fixing a crashing problem that you described in 5.3 I am running this on 1 phone and 5.3 on 3 others, the ones with 5.3 seem perfect, the one with 5.3.3 actually locked up once doing a transfer. Release 5.3.3: o GUI: fixed DND o GUI: fixed bug in displaying old voice mail messages o SIP: display local LED status for shared lines o WEB: + in settings value isn't anymore replaced by its hex value on settings dump web interface page o WEB: further enhanced french translation o SRTP: fixed bug with auto-answer o GUI: setting_server can be set manually via GUI menu (snom360) o GUI: ringer device should not switch to speaker if headset is enabled o GUI: dkeys (e.g. Redial, Retrieve) are working in edit number state, too o SETTINGS: if setting_server is IP:port only, make a valid URL out of it o SIP: display local LED status for shared lines o GUI: Shared Lines can be mapped to LEDs o LID: random number generated from random audio data http://fox.snom.com/download/snom360-5.3.3b-SIP-j.bin -Mike Mike240se -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Krief Sent: Friday, February 17, 2006 1:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Snom 360 Indeed - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 15, 2006 11:37 PM Subject: Re: [Asterisk-Users] Asterisk and Snom 360 On Wed, 15 Feb 2006, Olivier Krief wrote: Do not use 5.3 but 5.3.3 instead as major crashes occur with 5.3. http://www.snom.com/firmware.html#1641 5.3.3 is not available for public download... -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [slackware 10.2 and TE205P] Unknown signalling method 'pri_cpe'
On 2/19/06, Kevin Bockman [EMAIL PROTECTED] wrote: nik600 wrote: after some testing with [EMAIL PROTECTED], i've decided to install my asterisk server on a slackware (because it's my favourite distro and it is still suggested here http://www.voip-info.org/wiki-Asterisk+Linux+Slackware) , so, i've installed the last 10.2 release, and i've recompiled the 2.6.15.4 kernel. then i've downloaded asterisk1.2.4 and zaptel1.2.4 [chan_zap.so] = (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found Jan 2 00:16:57 ERROR[2059]: chan_zap.c:10606 setup_zap: Unknown signalling method 'pri_cpe' That's your problem. You don't have libpri installed. Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users sorry, i forget to say that i have installed libpri1.2.2 too installed with make make install ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Snom 360
Are you from snom? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Stredicke Sent: Sunday, February 19, 2006 6:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk and Snom 360 Still beta, but we could not make it crash any more...: We would be happy about the feedback from volunteers:-) http://fox.snom.com/download/snom320-5.3.6a-beta-SIP-j.bin http://fox.snom.com/download/snom320-5.3.6b-beta-SIP-j.bin http://fox.snom.com/download/snom360-5.3.6a-beta-SIP-j.bin http://fox.snom.com/download/snom360-5.3.6b-beta-SIP-j.bin Release 5.3.6: o LID: made sure audio channels are off in idle mode under all scenarios Release 5.3.5: o GUI: added cwi ringer indication o GUI: fixed unnecessary dialog state switches on shared line offhook o GUI: status led for missed calls o SIP: RAck in PRACK was buggy o SIP: added call pickup for shared lines Release 5.3.4: o SIP: added +sip.rendering parameter for BLA hold/resume NOTIFYs o SIP: NOTIFYs with subscription-state: terminated remove the subscription ~~~ Christian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Sunday, February 19, 2006 6:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk and Snom 360 Ok here it is, just remember who hooked you up :) But I don't see anything about fixing a crashing problem that you described in 5.3 I am running this on 1 phone and 5.3 on 3 others, the ones with 5.3 seem perfect, the one with 5.3.3 actually locked up once doing a transfer. Release 5.3.3: o GUI: fixed DND o GUI: fixed bug in displaying old voice mail messages o SIP: display local LED status for shared lines o WEB: + in settings value isn't anymore replaced by its hex value on settings dump web interface page o WEB: further enhanced french translation o SRTP: fixed bug with auto-answer o GUI: setting_server can be set manually via GUI menu (snom360) o GUI: ringer device should not switch to speaker if headset is enabled o GUI: dkeys (e.g. Redial, Retrieve) are working in edit number state, too o SETTINGS: if setting_server is IP:port only, make a valid URL out of it o SIP: display local LED status for shared lines o GUI: Shared Lines can be mapped to LEDs o LID: random number generated from random audio data http://fox.snom.com/download/snom360-5.3.3b-SIP-j.bin -Mike Mike240se -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Krief Sent: Friday, February 17, 2006 1:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Snom 360 Indeed - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 15, 2006 11:37 PM Subject: Re: [Asterisk-Users] Asterisk and Snom 360 On Wed, 15 Feb 2006, Olivier Krief wrote: Do not use 5.3 but 5.3.3 instead as major crashes occur with 5.3. http://www.snom.com/firmware.html#1641 5.3.3 is not available for public download... -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Snom 360
i would assume so, since his address is [EMAIL PROTECTED] -Dan On Sun, 19 Feb 2006, Michael J. Liberatore wrote: Are you from snom? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Stredicke Sent: Sunday, February 19, 2006 6:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk and Snom 360 Still beta, but we could not make it crash any more...: We would be happy about the feedback from volunteers:-) http://fox.snom.com/download/snom320-5.3.6a-beta-SIP-j.bin http://fox.snom.com/download/snom320-5.3.6b-beta-SIP-j.bin http://fox.snom.com/download/snom360-5.3.6a-beta-SIP-j.bin http://fox.snom.com/download/snom360-5.3.6b-beta-SIP-j.bin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application Faxing using SIP
I have successfully used the Grandstream ATA286 and Linksys PAP2NA. I would recommend the Grandstream over the Linksys as there is less configuration to do and it is IMHO more reliable for faxes. I have been able to get analog data modem connect at 48k on the grandstream whilst cannot get modem to work at all on Linksys. Craig - Original Message - From: Phil Blundell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 20, 2006 12:10 AM Subject: Re: [Asterisk-Users] Application Faxing using SIP On Sun, 2006-02-19 at 07:36 -0800, Lee Howard wrote: This will work provided that you can create a near-lossless communication path between the ATA and the PSTN gateway (which is the Asterisk box, I assume). One way of creating that, I would expect, would be to add another ethernet card to your Asterisk server and then run a crossover cable between that interface and the ethernet interface of the ATA. You'll also need to configure the ATA to not do lots of things typically done by ATAs, like echo cancellation. That's a good idea. I hadn't thought of using a crossover cable and a dedicated card like that. (Though, that said, I suspect that the datapath through our regular network switches is probably close enough to lossless for this purpose as well.) Any recommendations as to which ATAs are suitable for this purpose? I don't remember seeing a way to disable echo cancellation on either the Grandstream or the Sipura ones that I have here. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp 0.0.2pre25
Yes, I have compiled and used rxfax succesfully on 1.0.9, 1.0.10 and 1.2.4 to receive from analog fax machines. I have never yet been able to get rxfax working with txfax - my debugs when I try look like the logs in your email. Craig - Original Message - From: Jesse Guardiani [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, February 20, 2006 1:00 AM Subject: [Asterisk-Users] spandsp 0.0.2pre25 Hello, Is anyone successfully using spandsp 0.0.2pre25 with either asterisk 1.0.x or 1.2.4? I've built a Gentoo ebuild for this version of spandsp and app_rtxfax, and it builds, but I'm not having any luck getting it working. 99% of my test faxes fail. Reverting to 0.0.2pre20 yields a much higher success rate. I've bumped the console debugging level in logger.conf to include debug and verbose, as well as the defaults. I'm running libtiff-3.7.1, because Mr. Underwood's site recommends it, even though it's a vulnerable version of libtiff. I'm faxing from a working, production asterisk 1.0.9 + spandsp 0.0.2pre20 system to a testing asterisk 1.0.10 + spandsp 0.0.2pre25 system, and 1 of 3 things usually happens: 1.) The fax goes through (very rare in testing) 2.) The fax loops indefinitely like this: Feb 19 11:46:04 DEBUG[5089]: chan_zap.c:4059 zt_read: DTMF digit: f on Zap/1-1 Feb 19 11:46:04 DEBUG[5089]: chan_zap.c:4101 zt_read: Fax already handled Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: chan_zap.c:4059 zt_read: DTMF digit: f on Zap/1-1 Feb 19 11:46:07 DEBUG[5089]: chan_zap.c:4101 zt_read: Fax already handled Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10
[Asterisk-Users] Asterisk compile error
Hi Guys I have a problem compiling Asterisk 1.2.4. I am getting this error make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Has anyone come across this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [slackware 10.2 and TE205P] Unknown signalling method 'pri_cpe'
Please trim your responses; there is no need to quote the entire message including irrelevant text and signature lines. On Sunday 19 February 2006 18:25, nik600 wrote: sorry, i forget to say that i have installed libpri1.2.2 too installed with make make install Before or after you compiled zaptel and asterisk? It needs to be installed before you build everything else. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compile error
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 MBIT Technologies wrote: Hi Guys I have a problem compiling Asterisk 1.2.4. I am getting this error make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Has anyone come across this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You should provide more information, paste a few more lines from the breakage, look specifically for lines beginning with the word error. - -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFD+QzQEzF+JcQGyNIRAvhIAJwOIfxnmWRQ6kpPg2/7pRyqyklJhQCgl7LI 5B+xkAiZOTB2s/HqHFWkNmY= =QVhX -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How do I install speex for asterisk?
I used yum to install speex (although you could quite easily build your own from source). # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start This sequence of commands may require variation depending on your flavour of Linux and how your asterisk is installed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus E Zepeda Sent: Friday, 17 February 2006 10:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How do I install speex for asterisk? Huuu! I never expected you had to recompile asterisk to add a codec. But if that is what it takes, we'll do it. I noticed that asterisk makes reference to some speex.c in the makefile file. In some of those references I saw the actual speex.c file in the paths specified. A couple of them missing by the way. That could be why speex was never taken by asterisk. Mike, does speex have to be copied to a specific directory, then compiled and installed before re-compiling and re-installing asterisk? I appreciate you took your time to reply. Regards, Jesus -Original Message- From: Mike Pollitt [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 15:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How do I install speex for asterisk? You need to recompile Asterisk itself after installing Speex. Do a make clean, make, make install. I usually stop asterisk before that last step, by the way! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus E Zepeda Sent: Friday, 17 February 2006 5:58 AM To: Asterisk User List Subject: [Asterisk-Users] How do I install speex for asterisk? Hi, everybody: I enabled speex in my asterisk box (iax.conf), but no phone call went throug. At the asterisk console, I used the show modules command and it did not show the speex codec in the list. So, I downloaded the speex codec from speex.org, v1.0.5, compiled and installed in my asterisk machine. What I still don't know is: what do I need to do from the asterisk side to make it available? I just downloaded it to a directory, compiled and installed thinking that by doing a restart to asterisk it would some how know where to load it from. Any hints are appreciated Regards, Jesus E. Zepeda ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Viking CPC-Disconnect
Someone on the list a while back suggested that if you were having problems with call disconnects, to look into a product from Viking TellecomSolutions called cpc-disconnect: http://www.vikingtelecomsolutions.com/catalog/model_CPC-1.htm I received my unit on Friday and put it into place Saturday afternoon (SBC in this area doesn't supply call disconnect supervision). The unit was acting correctly, according to the instructions, but Asterisk never registered a disconnect. Is anybody using this unit? And, if so, can you share your setup? I'm currently using an Adit 600 channel bank, the cpc is sitting between the phone line and the Adit. Thanks! Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's
I've been using voicepulce connect for several months with very few problems. Occasionally I get "all circuits are busy" messages when trying to dial out but no too often. d From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. LiberatoreSent: Sunday, February 19, 2006 4:55 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's I had voicepulse connect but had to transfer IAX2 had non stop drop outs in audio all the time. Tried everything to fix it, even with 14ms ping times it just didnt want to work right. I never figured out why, just canceled. Although i didnt like the no-name on incoming caller id either though, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of andrew matthewsSent: Tuesday, February 14, 2006 8:52 PMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's http://connect.voicepulse.netThey support astrisk, with iax2 :) On 2/14/06, Jim Robinson [EMAIL PROTECTED] wrote: Hi Folks,Can anyone give me some good recommendations for VoIP providrs thatsupport Asterisk PBX's?We're based in Georgia and I having a hard timefinding anyoneRegards,JimPS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Loops and Variables
I have the following in my dialplan, counts the number of loops and when
it hits greater then 5, exit. It works, but errors initially with,
syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_LP or
tolken; Input: +1.
Could somebody tell me why?
Thanks:
;
; Setup a varriable to count the number of
; times the message has been played, when
; $COUNT reaches 5, play you've taken
; to long to dial and hangup.
;
exten = t,1,Set(COUNT=$[${COUNT} + 1])
exten = t,2,NoOP(${COUNT})
exten = t,3,GotoIf($[ ${COUNT} 5 ]?103)
exten = t,4,Goto(voice-mail-callback,s,4)
exten = t,103,Playback(local/tolong-todial)
exten = t,104,Playback(goodbye)
exten = t,105,Hangup()
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Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's
I have voicepulse connect too. I had occassional problems with incoming calls, but not many and not recently. Have had more problems with outgoing calls which is fine for me, as I have more than one backup (I use voxee as my primary due to lowest price, then voicepulse, and failing that I can use my cellphone or my landline). I am a bit disappointed with the price, it was decent before they upped it to $11. Seems a bit high to me, for just an incoming line with no outgoing minutes. Many other places charge about that and give you a bunch of minutes, or an unlimited local calling plan (in-state, in-area code, etc.). But, it's been very reliable, no complaints about uptime. Joseph Tanner On 2/19/06, David Blomquist [EMAIL PROTECTED] wrote: I've been using voicepulce connect for several months with very few problems. Occasionally I get all circuits are busy messages when trying to dial out but no too often. d From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Sunday, February 19, 2006 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's I had voicepulse connect but had to transfer IAX2 had non stop drop outs in audio all the time. Tried everything to fix it, even with 14ms ping times it just didnt want to work right. I never figured out why, just canceled. Although i didnt like the no-name on incoming caller id either though, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of andrew matthews Sent: Tuesday, February 14, 2006 8:52 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's http://connect.voicepulse.net They support astrisk, with iax2 :) On 2/14/06, Jim Robinson [EMAIL PROTECTED] wrote: Hi Folks, Can anyone give me some good recommendations for VoIP providrs that support Asterisk PBX's? We're based in Georgia and I having a hard time finding anyone Regards, Jim PS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loops and Variables
On Sun, 2006-02-19 at 20:30 -0500, Doug Lytle wrote: I have the following in my dialplan, counts the number of loops and when it hits greater then 5, exit. It works, but errors initially with, syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_LP or tolken; Input: +1. Could somebody tell me why? is count defined before it tries to do count + 1? if count is null you will see a parser error like that becuase it evaluates to count = + 1 -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loops and Variables
trixter aka Bret McDanel wrote: Could somebody tell me why? is count defined before it tries to do count + 1? No it isn't, thank you for the clue. I'll define it. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Cisco 7905 can't register
Mike Newton wrote:
My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k
on Debian stable). It could, however, register with another
installation of Asterisk and the settings on the phone (apart from the
SIP proxy address) haven't changed since then.
I was having this very problem last week; perhaps you've got the same issue.
You should make sure if you're behind NAT that your SIPDefault.cnf has the 2
NAT settings enabled. I didn't figure it out right away, because I was
focusing on the 401 unauthorized message, which doesn't really indicate a
NAT problem. I can't remember what the 2 settings are offhand, but they're
in the documentation.
I should be able to do this using the Web interface too, right, if I
disable TFTP downloading of the configuration? We have:
UID jeremy
PWD SECRET
Proxy tardis.malcolm.id.au
AltProxyTimeOut 0
UseLoginID 1
LoginID jeremy
SIPRegInterval 3600
MaxRedirect 5
SIPRegOn 1
NATIP 0.0.0.0
SIPPort 5060
MediaPort 16384
OutBoundProxy 0
NatServer 0
NatTimer 0x
DialPlan *St4-|#St4-|911|1#t8.r9t2-|0#t811.rat4-|^1t4#.-
IPDialPlan 1
But these are the same settings it had when it was registering to a
different Asterisk server...?
--
JEREMY MALCOLM [EMAIL PROTECTED] - lawyer, IT consultant and actor.
Internet and Open Source specialist. Web site: http://www.malcolm.id.au.
host -t NAPTR 1.0.8.0.3.1.2.9.8.1.6.e164.org |awk '{print substr($7,6)}'
smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [Asterisk-Users] Wildfire messsaging server
No timing issues. but we don't have much load anyway - it's sort of a small biz setup. but it sure looks like it can scale well. haven't used other IM platforms... Dean Collins wrote: Any issues with timing etc? up until now I've been very careful to run my asterisk as a standalone solution. Any issues with interfacing into other IM platforms? I'm happy to try this out but I have one client who uses MSN IM for voice a lot (I keep trying to get him to adopt skype but not interested). Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Fusion @ Gyantec Sent: Sunday, 19 February 2006 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Wildfire messsaging server Yes to both. It works perfectly fine - followed the instructions to the t; a few things you learn as you install :) let me know if you need any assistance or if you want us to install it for you rajeev Dean Collins wrote: http://www.jivesoftware.org/ Is anyone running a wildfire messaging server on the same pc as their asterisk server? Is anyone specifically running it on an [EMAIL PROTECTED] installation? TIA, Dean --- - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi setting ${DNIS}
Is there a reason the variable ${DNIS} does not get set with incoming
calls via chan_capi ?
Is it related to the MSN=X in capi.conf ?
version = chan_capi-cm-0.6.3
example;
exten = _9555XX,1,NoOp, ${EXTEN}, ${DNIS}
== ISDN1: Incoming call '04' - '9555'
-- Executing SetCDRUserField(CAPI/ISDN1/95 55-135, Incoming)
in new stack
-- Executing NoOp(CAPI/ISDN1/9555-135, 9555, ) in new stack
8 **SNIP**
CAPI INFO 0x3490: Normal call clearing
== Spawn extension (main, s, 2) exited non-zero on 'CAPI/
ISDN1/9555-135'
== ISDN1: CAPI Hangingup
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Re: [Asterisk-Users] Loops and Variables
On Sun, 2006-02-19 at 21:05 -0500, Doug Lytle wrote: trixter aka Bret McDanel wrote: Could somebody tell me why? is count defined before it tries to do count + 1? No it isn't, thank you for the clue. I'll define it. since you have had a little time to play with this, was this the problem? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk start errors with TDM2413E
I get the following errors when starting Asterisk. == Parsing '/etc/asterisk/zapata.conf': Found Feb 19 21:14:35 WARNING[10440]: chan_zap.c:920 zt_open: Unable to specify channel 1: No such device Feb 19 21:14:35 ERROR[10440]: chan_zap.c:6860 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Feb 19 21:14:35 ERROR[10440]: chan_zap.c:10264 setup_zap: Unable to register channel '1' Feb 19 21:14:35 WARNING[10440]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Feb 19 21:14:35 WARNING[10440]: loader.c:554 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# Ouch ... error while writing audio data: : Broken pipe Software versions asterisk-1.2.3 asterisk-addons-1.2.1 asterisk-perl-0.08 asterisk-sounds-1.2.1 libpri-1.2.2 zaptel-1.2.4 Output from modprobes [EMAIL PROTECTED] asterisk]# modprobe -v zaptel insmod /lib/modules/2.6.14-1.1656_FC4smp/misc/zaptel.ko [EMAIL PROTECTED] asterisk]# modprobe -v wctdm24xxp install /sbin/modprobe --ignore-install wctdm24xxp /sbin/ztcfg insmod /lib/modules/2.6.14-1.1656_FC4smp/misc/wctdm24xxp.ko This takes at least 10 seconds to come back to a prompt ztcfg output [EMAIL PROTECTED] asterisk]# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) Channel 05: FXS Kewlstart (Default) (Slaves: 05) Channel 06: FXS Kewlstart (Default) (Slaves: 06) Channel 07: FXS Kewlstart (Default) (Slaves: 07) Channel 08: FXS Kewlstart (Default) (Slaves: 08) Channel 09: FXS Kewlstart (Default) (Slaves: 09) Channel 10: FXS Kewlstart (Default) (Slaves: 10) Channel 11: FXS Kewlstart (Default) (Slaves: 11) Channel 12: FXS Kewlstart (Default) (Slaves: 12) Channel 13: FXS Kewlstart (Default) (Slaves: 13) Channel 14: FXS Kewlstart (Default) (Slaves: 14) Channel 15: FXS Kewlstart (Default) (Slaves: 15) Channel 16: FXS Kewlstart (Default) (Slaves: 16) 16 channels configured. zaptel.conf fxoks=1-4 fxsks=5-16 defaultzone=us loadzone=us zapata.conf [channels] signalling=fxo_ks echocancel=yes echocancelwhenbridged=yes usecallerid=yes context=outstation channel= 1-4 signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes usecallerid=yes group=2 context=incomingpstn channel= 5-16 Best regards, Duane Pudenz Network Infrastructure Manager Shasta Industries___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call forward on unavailable timer issues
I have a pretty standard
setup with Asterisk acting as a PABX for a bunch of SIP handsets (in this case,
SwissVoice IP10S).
My users are complaining
that when they forward their phones to their cellphones on unavailable (i.e.
forward when no-answer), their cellphone only rings once or twice, and then
Asterisk sends the call through to Voicemail.
Im using the
standard extension Macro thus:
[macro-stdexten]
; ${ARG1} - Extension
(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to
ring
; ${ARG3} - Voicemail
context
exten = s,1,Dial(${ARG2},20)
; Ring the interface, 20 seconds maximum
exten =
s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten =
s-NOANSWER,1,Voicemail([EMAIL PROTECTED]) ; If unavailable, send to
voicemail w/ unavail announce
exten =
s-BUSY,1,Voicemail([EMAIL PROTECTED]) ; If busy, send to voicemail w/
busy announce
exten =
_s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten = a,1,VoicemailMain([EMAIL PROTECTED])
; If they press *, send the user into VoicemailMain
Now clearly my problem is
that when the Dial application gets back a Temporarily Moved response from the
SIP phone (after the users preset period to wait before no-answer
forwarding), and drops back into the dialplan as Local/forwarded
number, the 20 second timer on the Dial command is still active.
I think what I need is a
way to reset or cancel this timer when a Temporarily Moved response comes back
in.
Surely this must be a
fairly common problem does anyone have a solution?
Thanks!
Mike.
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Re: [Asterisk-Users] chan_capi setting ${DNIS}
On 2/20/06, Nathan Alberti [EMAIL PROTECTED] wrote:
Is there a reason the variable ${DNIS} does not get set with incoming
calls via chan_capi ?
Is it related to the MSN=X in capi.conf ?
Just a guess, are you thinking of ${DNID} instead? There's no direct
mention of ${DNIS} on the wiki variables page, but ${DNID} works for
me with a BRI...
Andrew
--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
-- Bill Garrett
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RE: [Asterisk-Users] RE: Cisco 7905 can't register
Try this:
Use login ID: 0
Clear the Login ID Field so it's blank
lawyer, IT consultant and actor
Versatile us Aussies :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Malcolm
Sent: Monday, 20 February 2006 1:57 PM
To: Mike Newton
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] RE: Cisco 7905 can't register
Mike Newton wrote:
My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k
on Debian stable). It could, however, register with another
installation of Asterisk and the settings on the phone (apart from the
SIP proxy address) haven't changed since then.
I was having this very problem last week; perhaps you've got the same
issue.
You should make sure if you're behind NAT that your SIPDefault.cnf has the
2
NAT settings enabled. I didn't figure it out right away, because I was
focusing on the 401 unauthorized message, which doesn't really indicate a
NAT problem. I can't remember what the 2 settings are offhand, but
they're
in the documentation.
I should be able to do this using the Web interface too, right, if I
disable TFTP downloading of the configuration? We have:
UID jeremy
PWD SECRET
Proxy tardis.malcolm.id.au
AltProxyTimeOut 0
UseLoginID 1
LoginID jeremy
SIPRegInterval 3600
MaxRedirect 5
SIPRegOn 1
NATIP 0.0.0.0
SIPPort 5060
MediaPort 16384
OutBoundProxy 0
NatServer 0
NatTimer 0x
DialPlan *St4-|#St4-|911|1#t8.r9t2-|0#t811.rat4-|^1t4#.-
IPDialPlan 1
But these are the same settings it had when it was registering to a
different Asterisk server...?
--
JEREMY MALCOLM [EMAIL PROTECTED] - lawyer, IT consultant and actor.
Internet and Open Source specialist. Web site: http://www.malcolm.id.au.
host -t NAPTR 1.0.8.0.3.1.2.9.8.1.6.e164.org |awk '{print substr($7,6)}'
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[Asterisk-Users] Re: spandsp 0.0.2pre25
Craig Guy cguy at bigpond.net.au writes: Yes, I have compiled and used rxfax succesfully on 1.0.9, 1.0.10 and 1.2.4 to receive from analog fax machines. I have never yet been able to get rxfax working with txfax - my debugs when I try look like the logs in your email. Craig Perhaps I'm just being nitpicky, but you don't mention what version of spandsp you're using. pre20 rtfax - pre20 rxfax works fine here with asterisk 1.0.10 and 1.2.4. I tried using an analog fax machine with pre25 and asterisk 1.2.4 with no luck whatsoever. Unfortunately, I don't have the debug output from those attempts, but I could generate some if it would help. Jesse ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue Messages now playing when caller is inside queue
Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using Playback with this stanza: exten = 900,1,Playback(queue-youarenext) exten = 900,2,Playback(queue-thereare) exten = 900,3,Playback(digits/three) exten = 900,4,Playback(queue-callswaiting) exten = 900,5,Playback(vm-ivr) The queue is invoked by: exten = s,1,Background(Welcome) exten = s,2,Queue(callcenter|tT|||300) extern = s,3,Hangup When I tried exten = s,2,Queue(callcenter|tTr|||300) It was ringing with out music on hold, but again with out any announcement. Queue.conf is: [general] [default] [callcenter] music=default leavewhenempty = yes monitor-format = wav strategy=rrmemory timeout=15 retry=5 servicelevel = 60 wrapuptime=5 eventwhencalled = yes eventmemberstatusoff = no maxlen = 0 announce-frequency = 120 announce-holdtime = yes queue-thankyou = vm-ivr queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare ; (You are Currently caller no) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) context=vm member = Agent/1000 member = Agent/1001 member = Agent/1002 member = Agent/1003 member = Agent/1004 member = Agent/1005 The funny part is that it's working perfectly in the old setup. Did I make some mistake some where? I am running on debian stable and asterisk was compiled with simple make;make install. raj ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
Hello, world! I'm considering running Asterisk 1.2.4 on Solaris 10 on a Sun Fire X2100 server or two (Opteron CPU, nForce 4 chipset), and apparently this works. I've read that the Zaptel package won't work on anything other than Linux, since it's intended to hook into the Linux kernel in the form of a kernel module. This concerns me, since I've read that ztdummy, the timing-source component of Zaptel, is required for the music-on-hold and conferencing functions of Asterisk to function. So, with this in mind, is there any way to run a complete Asterisk solution on Solaris 10 (including music-on-hold and conferencing)? If so, how? Thanks in advance! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue Messages now playing when caller is insidequeue
Don't you need an exten = s,1,Answer ??? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rajkumar S Sent: Monday, 20 February 2006 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Queue Messages now playing when caller is insidequeue Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using Playback with this stanza: exten = 900,1,Playback(queue-youarenext) exten = 900,2,Playback(queue-thereare) exten = 900,3,Playback(digits/three) exten = 900,4,Playback(queue-callswaiting) exten = 900,5,Playback(vm-ivr) The queue is invoked by: exten = s,1,Background(Welcome) exten = s,2,Queue(callcenter|tT|||300) extern = s,3,Hangup When I tried exten = s,2,Queue(callcenter|tTr|||300) It was ringing with out music on hold, but again with out any announcement. Queue.conf is: [general] [default] [callcenter] music=default leavewhenempty = yes monitor-format = wav strategy=rrmemory timeout=15 retry=5 servicelevel = 60 wrapuptime=5 eventwhencalled = yes eventmemberstatusoff = no maxlen = 0 announce-frequency = 120 announce-holdtime = yes queue-thankyou = vm-ivr queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare ; (You are Currently caller no) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) context=vm member = Agent/1000 member = Agent/1001 member = Agent/1002 member = Agent/1003 member = Agent/1004 member = Agent/1005 The funny part is that it's working perfectly in the old setup. Did I make some mistake some where? I am running on debian stable and asterisk was compiled with simple make;make install. raj ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: Which ATA device do you recommend?
Michael J. Liberatore wrote: Are the PAP2's you can get branded vonage at staples for free after rebate still hackable? I read that you cant do it beyond a certain firmware but wasn't sure if it had to be connected to the internet for that download or if it ships with that now My understanding is that Cisco/Linksys now sells PAP2-NA without restriction. My suggestion was that the poster buy them. VoipSupply is a good source. I'm afraid I don't know about the branded ones. /edg ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Messages not playing when caller is inside queue
David Ankers wrote: Don't you need an exten = s,1,Answer The full sequence is: [ivr] ; Voice Menu exten = s, 1, wait(2) exten = s, 2, Answer exten = s, 3,Goto,MainMenu|s|1 [MainMenu] exten = s,1,Background(Welcome) exten = s,2,Queue(callcenter|tT|||600) extern = s,3,Hangup I am sorry that I missed this. The call is getting picked up and it goes to the agent in the queue. That part is fine. The only thing missing is that the messages (like queue-youarenext, queue-thankyou) are not played upon entering the queue. raj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rajkumar S Sent: Monday, 20 February 2006 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Queue Messages now playing when caller is insidequeue Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using Playback with this stanza: exten = 900,1,Playback(queue-youarenext) exten = 900,2,Playback(queue-thereare) exten = 900,3,Playback(digits/three) exten = 900,4,Playback(queue-callswaiting) exten = 900,5,Playback(vm-ivr) The queue is invoked by: exten = s,1,Background(Welcome) exten = s,2,Queue(callcenter|tT|||300) extern = s,3,Hangup When I tried exten = s,2,Queue(callcenter|tTr|||300) It was ringing with out music on hold, but again with out any announcement. Queue.conf is: [general] [default] [callcenter] music=default leavewhenempty = yes monitor-format = wav strategy=rrmemory timeout=15 retry=5 servicelevel = 60 wrapuptime=5 eventwhencalled = yes eventmemberstatusoff = no maxlen = 0 announce-frequency = 120 announce-holdtime = yes queue-thankyou = vm-ivr queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare ; (You are Currently caller no) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) context=vm member = Agent/1000 member = Agent/1001 member = Agent/1002 member = Agent/1003 member = Agent/1004 member = Agent/1005 The funny part is that it's working perfectly in the old setup. Did I make some mistake some where? I am running on debian stable and asterisk was compiled with simple make;make install. raj ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Messages not playing when caller is inside queue
In queues.conf: [queuename] announce-frequency = XX ; where XX = number of seconds Rajkumar S wrote: David Ankers wrote: Don't you need an exten = s,1,Answer The full sequence is: [ivr] ; Voice Menu exten = s, 1, wait(2) exten = s, 2, Answer exten = s, 3,Goto,MainMenu|s|1 [MainMenu] exten = s,1,Background(Welcome) exten = s,2,Queue(callcenter|tT|||600) extern = s,3,Hangup I am sorry that I missed this. The call is getting picked up and it goes to the agent in the queue. That part is fine. The only thing missing is that the messages (like queue-youarenext, queue-thankyou) are not played upon entering the queue. raj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rajkumar S Sent: Monday, 20 February 2006 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Queue Messages now playing when caller is insidequeue Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using Playback with this stanza: exten = 900,1,Playback(queue-youarenext) exten = 900,2,Playback(queue-thereare) exten = 900,3,Playback(digits/three) exten = 900,4,Playback(queue-callswaiting) exten = 900,5,Playback(vm-ivr) The queue is invoked by: exten = s,1,Background(Welcome) exten = s,2,Queue(callcenter|tT|||300) extern = s,3,Hangup When I tried exten = s,2,Queue(callcenter|tTr|||300) It was ringing with out music on hold, but again with out any announcement. Queue.conf is: [general] [default] [callcenter] music=default leavewhenempty = yes monitor-format = wav strategy=rrmemory timeout=15 retry=5 servicelevel = 60 wrapuptime=5 eventwhencalled = yes eventmemberstatusoff = no maxlen = 0 announce-frequency = 120 announce-holdtime = yes queue-thankyou = vm-ivr queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare ; (You are Currently caller no) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) context=vm member = Agent/1000 member = Agent/1001 member = Agent/1002 member = Agent/1003 member = Agent/1004 member = Agent/1005 The funny part is that it's working perfectly in the old setup. Did I make some mistake some where? I am running on debian stable and asterisk was compiled with simple make;make install. raj ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp 0.0.2pre25
pre25 is working fine for me. On 2/19/06, Jesse Guardiani [EMAIL PROTECTED] wrote: Hello, Is anyone successfully using spandsp 0.0.2pre25 with either asterisk 1.0.x or 1.2.4? I've built a Gentoo ebuild for this version of spandsp and app_rtxfax, and it builds, but I'm not having any luck getting it working. 99% of my test faxes fail. Reverting to 0.0.2pre20 yields a much higher success rate. I've bumped the console debugging level in logger.conf to include debug and verbose, as well as the defaults. I'm running libtiff-3.7.1, because Mr. Underwood's site recommends it, even though it's a vulnerable version of libtiff. I'm faxing from a working, production asterisk 1.0.9 + spandsp 0.0.2pre20 system to a testing asterisk 1.0.10 + spandsp 0.0.2pre25 system, and 1 of 3 things usually happens: 1.) The fax goes through (very rare in testing) 2.) The fax loops indefinitely like this: Feb 19 11:46:04 DEBUG[5089]: chan_zap.c:4059 zt_read: DTMF digit: f on Zap/1-1 Feb 19 11:46:04 DEBUG[5089]: chan_zap.c:4101 zt_read: Fax already handled Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: chan_zap.c:4059 zt_read: DTMF digit: f on Zap/1-1 Feb 19 11:46:07 DEBUG[5089]: chan_zap.c:4101 zt_read: Fax already handled Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down
RE : [Asterisk-Users] Asterisk start errors with TDM2413E
Title: Message Hi, I believe that you have inverted fxo_ks and fxs_ks into your zapata.cong file "signaling=" declaration... Invert and redo the tests. Good Luck ! Francois BERGERET, France. -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED]Envoyé: lundi 20 février 2006 04:34À: asterisk-users@lists.digium.comObjet: [Asterisk-Users] Asterisk start errors with TDM2413EI get the following errors when starting Asterisk. == Parsing '/etc/asterisk/zapata.conf': Found Feb 19 21:14:35 WARNING[10440]: chan_zap.c:920 zt_open: Unable to specify channel 1: No such device Feb 19 21:14:35 ERROR[10440]: chan_zap.c:6860 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Feb 19 21:14:35 ERROR[10440]: chan_zap.c:10264 setup_zap: Unable to register channel '1' Feb 19 21:14:35 WARNING[10440]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Feb 19 21:14:35 WARNING[10440]: loader.c:554 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# Ouch ... error while writing audio data: : Broken pipe Software versions asterisk-1.2.3 asterisk-addons-1.2.1 asterisk-perl-0.08 asterisk-sounds-1.2.1 libpri-1.2.2 zaptel-1.2.4 Output from modprobes [EMAIL PROTECTED] asterisk]# modprobe -v zaptel insmod /lib/modules/2.6.14-1.1656_FC4smp/misc/zaptel.ko [EMAIL PROTECTED] asterisk]# modprobe -v wctdm24xxp install /sbin/modprobe --ignore-install wctdm24xxp /sbin/ztcfg insmod /lib/modules/2.6.14-1.1656_FC4smp/misc/wctdm24xxp.ko This takes at least 10 seconds to come back to a prompt ztcfg output [EMAIL PROTECTED] asterisk]# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) Channel 05: FXS Kewlstart (Default) (Slaves: 05) Channel 06: FXS Kewlstart (Default) (Slaves: 06) Channel 07: FXS Kewlstart (Default) (Slaves: 07) Channel 08: FXS Kewlstart (Default) (Slaves: 08) Channel 09: FXS Kewlstart (Default) (Slaves: 09) Channel 10: FXS Kewlstart (Default) (Slaves: 10) Channel 11: FXS Kewlstart (Default) (Slaves: 11) Channel 12: FXS Kewlstart (Default) (Slaves: 12) Channel 13: FXS Kewlstart (Default) (Slaves: 13) Channel 14: FXS Kewlstart (Default) (Slaves: 14) Channel 15: FXS Kewlstart (Default) (Slaves: 15) Channel 16: FXS Kewlstart (Default) (Slaves: 16) 16 channels configured. zaptel.conf fxoks=1-4 fxsks=5-16 defaultzone=us loadzone=us zapata.conf [channels] signalling=fxo_ks echocancel=yes echocancelwhenbridged=yes usecallerid=yes context=outstation channel= 1-4 signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes usecallerid=yes group=2 context=incomingpstn channel= 5-16 Best regards,Duane PudenzNetwork Infrastructure ManagerShasta Industries ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Messages not playing when caller is inside queue
Peter Fern wrote: In queues.conf: [queuename] announce-frequency = XX ; where XX = number of seconds I had already given it. From my orig mail: [callcenter] music=default leavewhenempty = yes monitor-format = wav strategy=rrmemory timeout=15 retry=5 servicelevel = 60 wrapuptime=5 eventwhencalled = yes eventmemberstatusoff = no maxlen = 0 announce-frequency = 120 announce-holdtime = yes queue-thankyou = vm-ivr queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare ; (You are Currently caller no) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) context=vm member = Agent/1000 member = Agent/1001 member = Agent/1002 member = Agent/1003 member = Agent/1004 member = Agent/1005 Again this config is working perfectly in the 1.0.9-BRIstuffed-0.2.0-RC8h (Xorcom Rapid), but not in 1.2.4 raj ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Cisco 7905 can't register
David Ankers wrote:
Try this:
Use login ID: 0
Clear the Login ID Field so it's blank
Thanks but no, already tried that and no difference. I also tried three
other different versions of Asterisk: 1.2.1, 1.2.4 and on a whim
downgrading to 1.0.2, and tried the conf files from the old installation
of Asterisk with which I could successfully register, none of which helped.
I'm running it on a UML (User Mode Linux) machine. Would that make any
difference?
I still get:
-- SIP read from 220.238.251.245:50205:
REGISTER sip:tardis.malcolm.id.au SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:5060
From: sip:[EMAIL PROTECTED];tag=2558233792
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
Contact: Jeremy Malcolm
sip:[EMAIL PROTECTED]:5060;transport=udp;expires=3600
User-Agent: Cisco-CP7905/1.01-030807A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
--- (10 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.0.104 : 5060 (NAT)
Transmitting (NAT) to 220.238.251.245:50205:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.104:5060;received=220.238.251.245
From: sip:[EMAIL PROTECTED];tag=2558233792
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
---
Transmitting (NAT) to 220.238.251.245:50205:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.104:5060;received=220.238.251.245
From: sip:[EMAIL PROTECTED];tag=2558233792
To: sip:[EMAIL PROTECTED];tag=as34ca76e7
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:[EMAIL PROTECTED]
WWW-Authenticate: Digest realm=malcolm.id.au, nonce=2bfd861a
Content-Length: 0
TIA
--
JEREMY MALCOLM [EMAIL PROTECTED] - lawyer, IT consultant and actor.
Internet and Open Source specialist. Web site: http://www.malcolm.id.au.
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[Asterisk-Users] Live Communication Server and Asterisk
Has anyone have interfaced this successfully? I came to know from M$ that Genesys GETS can be used to interface asterisk. I have interfaced Cisco call manager to asterisk/ser but for my final setup I would like to have a LCS talking to a CCM, without having the Genesys GETS is I dont have to. Has anyone been playing around with this? If so I would really like to hear some advise. Regards, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : [EMAIL PROTECTED] WWW: www.imcb.a-star.edu.sg Hi all,i have now managed to place a call from LCS toasterisk/pstn and it seems to work fine. Unfortunatelyi have still problems for incomming calls fromasterisk/pstn to LCS.i have seen in the mailinglist that there seems to beproblem calling from lcs to asterisk. Have anyonemaneged to place a call from lcs to *.thx in advance... --- richard Coco coco_richard at yahoo.com wrote: Hi, i have the same setup too. [exten_3008]-[asterisk/TCP_SUPPORT]-[LCS]-[exten_20] Unfortunately i don't know how to configure the dialplan in my LCS. Can you please give me a hint where to configure this. thx. --- Jacky jacky.tw at gmail.com wrote: LCS 2005 just support SIP TCP or TLS right now. so you must patch asterisk chan_sip.c support TCP, look http://bugs.digium.com/view.php?id=4903I have successful call to asterisk's SIP peer or PSTN use Office Communicator 2005(sign-in my LCS 2005) but I can't use Dial(SIP/username at lcs.domain) , let asterisk's SIP user invite LCS's user.Need any input. 2005/8/11, bubuk bubuk at ish.de: Hi List! does anyone played around with the LCS and Asterisk? Because the LCS is doing no RFC compliant SIP, i wonder if it can work. Google couldn't tell me. If someon heared about that, please let me know. The fact i figured out is that the Border Controler from Jasomi can be used as a gateway from MS-LCS-SIP to regular SIP. But that is not really handy and expensive too. Thank you Volker ___ Asterisk-Users mailing list Asterisk-Users at lists.digium.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Cisco 7905 can't register - SOLVED
Do you know what the problem was? It was that I had three default
routes (all identical). This affected nothing adversely except for
Asterisk.
So, there you go.
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JEREMY MALCOLM [EMAIL PROTECTED] - lawyer, IT consultant and actor.
Internet and Open Source specialist. Web site: http://www.malcolm.id.au.
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RE: [Asterisk-Users] Fwd: Which ATA device do you recommend?
Anyone have a great reference for configuring the PAP2-NA with Asterisk? -Original Message- From: Ed Greenberg [mailto:[EMAIL PROTECTED] Sent: Sunday, February 19, 2006 11:57 PM To: Michael J. Liberatore Cc: Asterisk User List Subject: Re: [Asterisk-Users] Fwd: Which ATA device do you recommend? Michael J. Liberatore wrote: Are the PAP2's you can get branded vonage at staples for free after rebate still hackable? I read that you cant do it beyond a certain firmware but wasn't sure if it had to be connected to the internet for that download or if it ships with that now My understanding is that Cisco/Linksys now sells PAP2-NA without restriction. My suggestion was that the poster buy them. VoipSupply is a good source. I'm afraid I don't know about the branded ones. /edg ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line Dropouts on E405P
On Mon, 20 Feb 2006, Asterisk - Mailing List wrote: We have a Ericsson BP250 Phone system setup witht he following configuration Telco - Asterisk E405P - BP250 The system seem to work perfectly on 1.0.9 for a very long time but there is some functionality we wanted to take advantage of in the 1.2 version branch so we upgraded. Currently running Asterisk 1.2.4 Zaptel 1.2.3 (noticed 1.2.4 is out will upgrade next weekend) Libpri 1.2.2 The problem we are getting is wierd but :- Sorry about the timings looking wierd but you have to allow a fudge factor of anywhere upto 12 hours when dealing with reports from on-site personel. * Wednesday ~ 9.30am All calls drops for 1 second, then back online, then ~5 minutes later, same thing * Thursday ~ 10.30am All calls drops for 1 second, then back online, then ~5 minutes later, same thing * Friday ~ 11.30am All calls drops for 1 second, then back online, then ~5 minutes later, same thing Hi, We have a client who had exactly this situation - 1.0.9 running stable, 1.2 falling over like you. In the end Digium support helped out. The problem was fixed by disabling the framebuffer in the system setup. Don't know why 1.0.9 and 1.2 were different. Still - there's a clue for you. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
Hey Alex, Please forgive the question, but what is the rationale behind using Solaris over Linux as an asterisk hosting platform? Cheers, Mark -Original Message- From: Alexander Burke [mailto:[EMAIL PROTECTED] Sent: Monday, 20 February 2006 3:45 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron,Sun Fire X2100) Hello, world! I'm considering running Asterisk 1.2.4 on Solaris 10 on a Sun Fire X2100 server or two (Opteron CPU, nForce 4 chipset), and apparently this works. I've read that the Zaptel package won't work on anything other than Linux, since it's intended to hook into the Linux kernel in the form of a kernel module. This concerns me, since I've read that ztdummy, the timing-source component of Zaptel, is required for the music-on-hold and conferencing functions of Asterisk to function. So, with this in mind, is there any way to run a complete Asterisk solution on Solaris 10 (including music-on-hold and conferencing)? If so, how? Thanks in advance! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
