Re: [asterisk-users] SSH connection hangs on logout?

2006-08-23 Thread Tzafrir Cohen
On Wed, Aug 23, 2006 at 11:03:23PM -0700, Steve Edwards wrote: > On Thu, 24 Aug 2006, Jeremy McNamara wrote: > > >Rushowr wrote: > >>Hey all, I have an interesting issue that just recently started when I > >>grabbed a copy of the trunk about a week ago and compiled it. Ever since > >>that compile,

[asterisk-users] AstLinux 0.4.3 Released!

2006-08-23 Thread Kristian Kielhofner
Hello everyone, I have released AstLinux 0.4.3: http://sourceforge.net/projects/astlinux/ For all of those that have been waiting to switch to 0.4.x, this is your chance. The few remaining problems with uclibc have been fixed (i.e. voicemail timezones and voicemail -> email via MSMT

Re: [asterisk-users] SSH connection hangs on logout?

2006-08-23 Thread Steve Edwards
On Thu, 24 Aug 2006, Jeremy McNamara wrote: Rushowr wrote: Hey all, I have an interesting issue that just recently started when I grabbed a copy of the trunk about a week ago and compiled it. Ever since that compile, if I start Asterisk (disconnected terminal, using safe_asterisk to launch) and

Re: [asterisk-users] Annoying Bristuff

2006-08-23 Thread Andrew Nowrot
HiThe newest bristuff didn't change anything. Still the same. I was wondering if this is happening only to me or not. Does anyone has the same problem? Maybe I am messing something when loading the modules. Does anyone have any other tips.Andrew ___ --Ban

Re: [asterisk-users] Nokia E60/61/70 and SIP

2006-08-23 Thread Dinesh Nair
On 08/24/06 09:02 El Flynn said the following: Just wondering -- has anyone used the SIP phone feature on the Nokia E60/61/70 phones? We're trying to see if this would be an OK phone to get for the company, particularly since we're already running Asterisk. SIP works well with asterisk, with

RE: [asterisk-users] Nokia E60/61/70 and SIP

2006-08-23 Thread Haspers
We are using some E61 and E70's with asterisk. Only problem we have at this moment is that we are unable to use a password for the authentication. I haven't found out yet why this isn't working. They are working good, but I would like to see some small things changed in future firmware versions (li

Re: [asterisk-users] Realtime Extensions -- Comments?

2006-08-23 Thread Brian Capouch
Douglas Garstang wrote: It doesn't matter where you turn in Asterisk, there's gotcha's. For example, you can't put the hint stuff into realtime, and there's no inherint way to comment extensions. It doesn't seem like Asterisk is good enough for you Doug. Switch to one of the competitors' p

Re: [asterisk-users] Nokia E60/61/70 and SIP

2006-08-23 Thread atik khan
Hi, have you looked here http://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html thanks atik On 8/24/06, El Flynn <[EMAIL PROTECTED]> wrote: Hi list, Just wondering -- has anyone used the SIP phone feature on the Nokia E60/61/70 phones? We're trying to see if this would be an OK phone t

Re: [asterisk-users] SSH connection hangs on logout?

2006-08-23 Thread Jeremy McNamara
Rushowr wrote: Hey all, I have an interesting issue that just recently started when I grabbed a copy of the trunk about a week ago and compiled it. Ever since that compile, if I start Asterisk (disconnected terminal, using safe_asterisk to launch) and then continue on about my work with it, when

[asterisk-users] SSH connection hangs on logout?

2006-08-23 Thread Rushowr
Hey all, I have an interesting issue that just recently started when I grabbed a copy of the trunk about a week ago and compiled it. Ever since that compile, if I start Asterisk (disconnected terminal, using safe_asterisk to launch) and then continue on about my work with it, when I disconnect my S

RE: [asterisk-users] NAT problems

2006-08-23 Thread Sergio R. D'Ippolito
Try changing the configuration on your PAP2 linksys, more precisly the part where is the NAT parameters, try changing the options from NO to YES. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de andrutto Enviado el: Miércoles, 23 de Agosto de 2006 03:41 p.m.

RE: [asterisk-users] Re: problems with wevbmail

2006-08-23 Thread Sergio R. D'Ippolito
I could fix it.   The problem was permissions on the  directory /var/spool/asterisk/voicemail.   Thanks     De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Steven Enviado el: Miércoles, 23 de Agosto de 2006 08:01 a.m. Para: asterisk-users@lists.digium.com Asunto:

Re: [asterisk-users] using asterisk + sangoma a102 to simulate telco PRI: is possible?

2006-08-23 Thread Rich Adamson
Giorgio Incantalupo wrote: Hi, I have an asterisk box with a sangoma a102 (two PRI ports). Is is possible to connect port A to port B in order to use port B as a simulation of a telco PRI line? If yes, is there a special cable needed? How can I configure the card and zaptel.conf? Yes. You'll

[asterisk-users] Nokia E60/61/70 and SIP

2006-08-23 Thread El Flynn
Hi list, Just wondering -- has anyone used the SIP phone feature on the Nokia E60/61/70 phones? We're trying to see if this would be an OK phone to get for the company, particularly since we're already running Asterisk. Not asking for a review of the phone, but rather how well the built-in SI

Re: [asterisk-users] About IVR and Oracle

2006-08-23 Thread Time Bandit
On 8/23/06, Infobox Peru <[EMAIL PROTECTED]> wrote: maybe you could make it with PHP and its driver for Oracle. For PHP have a look here : http://phpagi.sourceforge.net/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailin

[asterisk-users] SJPhone and Asterisk over H323

2006-08-23 Thread Matt King
Hello all, I'm using Asterisk h323 (default/NuPhone) with some success with SJPhone. I say some success because while I'm able to receive audio from Asterisk, I seem unable to send audio to it... Any suggestions? Anybody managed to get this to work? Thanks, Matt King

[asterisk-users] Getting strange behavior on SIP channels after upgrade to 1.2.11

2006-08-23 Thread Álvaro Palma
I upgraded to 1.2.11 and now I see two behaviors different than the existent in 1.2.10: 1.- I get 183 Session Progress instead of 180 Ringing. 2.- If I have three extensions, A, B and C. A using codec X, B using also codec X and C using codec Y. If C dials to B and A tries to pick up the call (usi

Re: [asterisk-users] Re: Asterisk IAXmodem HylaFax?

2006-08-23 Thread Warrick Zedi
Tzafrir Cohen wrote: On Wed, Aug 23, 2006 at 03:41:22PM +1000, Warrick Zedi wrote: Tzafrir, When last did you look at AsterFax? What do you believe is required to set it up? In what way are there "wheel reinventings" in either HylaFax or AsterFax? Tzafrir Cohen wrote: On Wed, Aug 2

Re: [asterisk-users] About IVR and Oracle

2006-08-23 Thread Infobox Peru
maybe you could make it with PHP and its driver for Oracle. Daniel Pizarro www.infobox-peru.com-- Forwarded message --From: Moises Silva < [EMAIL PROTECTED]>Date: 23-ago-2006 17:12Subject: Re: [asterisk-users] About IVR and OracleTo: Asterisk Users Mailing List - Non-Commercial Dis

[asterisk-users] One way audion on Sangoma

2006-08-23 Thread Dovid Bender
Hi List, I have an A200 with echo can. 2-FXO and 2 FXS. Today I went and upgraded asterisk, zaptel and libpri. I ran the sangoma util to patch asterisk. When I started up asterisk ZAP1 worked like a charm. However ZAP2 has been acting up. I only get one way audio on it. The person that I call

[asterisk-users] howto install asterisk on freebsd release 4.11

2006-08-23 Thread mansour safai
Hi There, Is there anybody who installed asterisk on freebsd 4.11 release ? I was not succesful. please guide me. I updated the ports and I installed the lib using ports but when I try to install zaptel it says cannot load it for release before than 5 I couldn't install the asterisk from ports als

RE: [asterisk-users] Registering IP Phone To Asterisk

2006-08-23 Thread David Gagnon
I don't think you can use the template of another brand with your Fanvil. You must configure the phone manually. First time I ear about FAnvil IP phone so I cannot help you David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED] Envoyé : 2

RE: [asterisk-users] Hint extension issue - bug?

2006-08-23 Thread David Gagnon
I tested the trunk two days ago and I agree that the presence feature is broken. I don't know if it's an error or if it's volunteer. I will post a bug report on the tracker and we will see. One thing is sure; hints are working well on 1.2.10 and not in the trunk. Is this because they did some chan

Re: [asterisk-users] About IVR and Oracle

2006-08-23 Thread Moises Silva
Sure is possible. Look into google 'asterisk agi fastagi'. Regards On 8/23/06, Javier Lara Sanchez <[EMAIL PROTECTED]> wrote: Dear All, I need to buid an IVR that could make a request to a data base (oracle) in a remote host. The idea is that an user dial a extension with 2 options a

[asterisk-users] About IVR and Oracle

2006-08-23 Thread Javier Lara Sanchez
Dear All,   I need to buid an IVR that could make a request to a data base (oracle) in a remote host.    The idea is that an user dial a extension with 2 options and one of them ask for a data (in the case a date). This data is the field that the data base needs to find the information

Re: [asterisk-users] USB GSM gateway for Asterisk?

2006-08-23 Thread Zoa
You can find cheap gsm (+/- 150$) gateways too, although the cheap ones will require a additional pstn card. (expensive ones could do sip) Zoa. Jay Milk wrote: There's been some (futile?) effort a while back attempting to get a Bluetooth capable phone integrated into asterisk as a channel.

[asterisk-users] isdn30 uk setup problem

2006-08-23 Thread Nicholas Colyer
Hi Can anyone help with my following problem connecting asterisk to a new provisioned isdn30e line. At long last I have had BT install our new isdn 30 (I421) line for our asterisk server after 3 months of waiting. Before this we have used asterisk with a tdm400 and some analogue lines. I

Re: [asterisk-users] Annoying Bristuff

2006-08-23 Thread Andrew Nowrot
HiThanks for your reply.I will check it first thing in a morning and of course will let you know about results.Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http:

Re: [asterisk-users] How to set externip in sip.conf automatically?

2006-08-23 Thread Larry Alkoff
John Marvin wrote: Larry Alkoff wrote: As stated in the original post, when I entter the IP with an editor directly into sip.conf calls work just fine but I am looking for a way to have that done _automatically_. The Asterisk - Future of Telephony book says it is possible for Asterisk to acc

[asterisk-users] NoCDR()

2006-08-23 Thread Doug Lytle
Everybody, What is the proper usage of NoCDR()? I keep getting the following warning about lacks end: Aug 23 16:34:32 WARNING[23822]: cdr.c:443 ast_cdr_free: CDR on channel 'Local/[EMAIL PROTECTED],1' not posted Aug 23 16:34:32 WARNING[23822]: cdr.c:445 ast_cdr_free: CDR on channel 'Local/[

Re: [asterisk-users] How to set externip in sip.conf automatically?

2006-08-23 Thread John Marvin
Larry Alkoff wrote: As stated in the original post, when I entter the IP with an editor directly into sip.conf calls work just fine but I am looking for a way to have that done _automatically_. The Asterisk - Future of Telephony book says it is possible for Asterisk to access a Linux environm

Re: [asterisk-users] How to set externip in sip.conf automatically?

2006-08-23 Thread Larry Alkoff
Thank you Greg and RR. externhost=myhost.dyndns.org works perfectly so figuring out how to access a shell variable from within the CLI is no longer necessary - although it would be nice to know! externhost works in 1.20 onwards. Thanks for finding the solution. Larry Greg Delgado wrote: T

[asterisk-users] MySQL undefined symbol: __pure_virtual

2006-08-23 Thread Dan Brummer
Hey guys, I'm getting the following message when I start asterisk:   Aug 23 13:42:40 WARNING[29258] loader.c: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: __pure_virtual Aug 23 13:42:40 WARNING[29258] loader.c: Loading module res_config_mysql.so failed!   I don't know ho

[asterisk-users] Unable to start special tone

2006-08-23 Thread Kevin Savoy
Can anyone tell me where this is coming from? I can’t seem to find any information on it anywhere. I don’t believe I’m using “special tones” anywhere. Any ideas?   Aug 23 14:30:34 WARNING[15443]: chan_zap.c:5492 ss_thread: Unable to start special tone on 15   _ Kevi

Re: [asterisk-users] AMI initiate call probs

2006-08-23 Thread Niklas Larsson
On Tue, 22 Aug 2006 16:55:37 +0200, Niklas Larsson wrote: > I'm using AMI to initiate a call, first calling the agent and when > he picks up, the call is placed to the customer. The prob is if the > user rejects the call (or they don't have cw...), the call is still > placed to the customer... > >

Re: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install

2006-08-23 Thread Doug Lytle
[EMAIL PROTECTED] wrote: Looking for a way to hard reset a ADIT 600 just purchased used. But it seems to have a master password already set. We've tried the front reset but maybe we don't have the right sequence of boot order. Any help would be much appreciated? - Jim Jim, Did you ever

Re: [asterisk-users] Annoying Bristuff

2006-08-23 Thread Tzafrir Cohen
On Wed, Aug 23, 2006 at 09:35:17PM +0200, Andrew Nowrot wrote: > Hi > > I am trying to set up * box with the ISDN hfc-s cards. One in NT mode and > two in TE. I am using the bristuff-0.3.0-PRE-1r.tar.gz . The installation > went well, but soon after the zaphfc was loaded I started to receive these

[asterisk-users] client socket to asterisk manager gets disconnected

2006-08-23 Thread Roi Stork
I have a test application, what it does is just connect to the asterisk manager, and listen for events. I also set the connection to receive on user, call and agent events. I Noticed that everytime the queue is empty and a caller joins in, asterisk tends to throw too many queuememberstatus event

Re: [asterisk-users] 3COM NBX and Digium Cards...

2006-08-23 Thread Antoine Megalla
--- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 189 bytes > Desc: This is a digitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20060823/486aabc3/attachment-0001

Re: [asterisk-users] Working Sipura 3000 or Linksys 3102 configuration?

2006-08-23 Thread Ron Wellsted
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 incent Delporte wrote: > Hello > > I'm having a problem with the Linksys 3102: With incoming PSTN > calls, I can hear the caller through the X-Ten softphone, but he can't > hear me. The problem is worse with Sjphone and the GrandStream 100 > hardph

[asterisk-users] Re: NAT problems

2006-08-23 Thread andrutto
Strange?!? These three phones are using g726 (this codec is configured in sip.conf and in SIP ATA as well). -- Zostan Dziewczyna Lata! >>> http://link.interia.pl/f1997 ___ --Bandwidt

[asterisk-users] Annoying Bristuff

2006-08-23 Thread Andrew Nowrot
HiI am trying to set up * box with the ISDN hfc-s cards. One in NT mode and two in TE. I am using the bristuff-0.3.0-PRE-1r.tar.gz . The installation went well, but soon after the zaphfc was loaded I started to receive these message in kernlog:Aug 23 21:00:08 asterisk kernel: zaphfc: empty HDLC fra

Re: [asterisk-users] NAT problems

2006-08-23 Thread Eric \"ManxPower\" Wieling
andrutto wrote: Hi, Does anyone know how to solve this issue. I have Asterisk box on public IP and three clients connected to it. Unfortunately they are behind NAT (simple one-to-one). Those three clients can make outgoing calls hassle free, but when I try to make a call between them somethi

Re: [asterisk-users] Direct to Voicemail

2006-08-23 Thread Doug Lytle
Aaron Daniel wrote: Since you're using the variables to decide what to do next (VMBOXEXISTSSTATUS), you can go ahead and set priorityjumping=no in the Thank you very much, this took care of it. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Te

Re: [asterisk-users] Dialplan "or" matching

2006-08-23 Thread Kevin Smith
Glad I could help. I agree, these mailing lists are a life saver. I personally have only been using Asterisk for about 5 months now, in fact I have never even delt with any PBX's before (complete newbie) but everyone here is very helpful and I am picking up a lot. Kevin David Cook wrote: Tha

[asterisk-users] NAT problems

2006-08-23 Thread andrutto
Hi, Does anyone know how to solve this issue. I have Asterisk box on public IP and three clients connected to it. Unfortunately they are behind NAT (simple one-to-one). Those three clients can make outgoing calls hassle free, but when I try to make a call between them something is not right.

Re: [asterisk-users] How to set externip in sip.conf automatically?

2006-08-23 Thread Larry Alkoff
That's a very nice idea Greg. I'm not sure that my Asterisk 1.2 has the externhost= function but it would solve my problem. I have a dyndns.org account already that reports my externip. Larry Greg Delgado wrote: The easiest way is to register for free dynamic DNS service at www.dyndns.com.

[asterisk-users] 3COM NBX and Digium Cards...

2006-08-23 Thread Carlos Chavez
I have a customer that has returned two cards, a TE210P and a TE110P because they are no longer working. Both cards were connected to an 3COM NBX system but not to the same one. On the TE210P only the port that was connected to the NBX failed, the other works perfectly. The car

[asterisk-users] Choppy calls on IAX trunk but no problems on internal calls

2006-08-23 Thread Carl Youngblood
We are running the default asterisk package on Ubuntu Dapper (which has the advanced timing options used by ztdummy). Our connection to the PSTN is over an IAX trunk with our provider. We are getting really bad call quality on calls over the IAX trunk--voice seems to be garbled or out of order a

[asterisk-users] USB GSM gateway for Asterisk?

2006-08-23 Thread Jay Milk
There's been some (futile?) effort a while back attempting to get a Bluetooth capable phone integrated into asterisk as a channel. The idea, of course, was to make it possible to have asterisk utilize a cellular connection for backup, calls on free nights/weekends, or free in-network minutes.

RE: [asterisk-users] Sipura SPA-3000 vs Sangoma A200

2006-08-23 Thread Mindaugas Kezys
Hello, Ok, few bad words about A200. Our company is based in Lithuania. Our company used SPA-3000, but because of echo problems we are not using them anymore. Now we are trying our luck with Sangoma A200 but the following problem occurred on few systems we installed. When calling person hangups

RE: [asterisk-users] Sipura SPA-3000 vs Sangoma A200

2006-08-23 Thread Mindaugas Kezys
Hello, Ok, few bad words about A200. Our company is based in Lithuania. Our company used SPA-3000, but because of echo problems we are not using them anymore. Now we are trying our luck with Sangoma A200 but the following problem occurred on few systems we installed. When calling person hangups

[asterisk-users] IAX2 extn not registering on 4569

2006-08-23 Thread [EMAIL PROTECTED]
Hi all,   Just having a strange situation with no clues how to solve.   I have an Asterisk/TRIXBOX located in US and an IAX extn running on PA168V ATA in another country. All my configs seems to be on 4569 but i see my extn connected at a different port like 13569.   How can i make it to register

[asterisk-users] Choppy calls on IAX trunk but no problems on internal calls

2006-08-23 Thread Carl Youngblood
We are running the default asterisk package on Ubuntu Dapper. Our connection to the PSTN is over an IAX trunk with our provider. We are getting really bad call quality on calls over the IAX trunk--voice seems to be garbled or out of order and often completely breaks up. But on internal calls bet

[asterisk-users] Silent Calls (Ghost Calls) When Picking Up Queue Calls

2006-08-23 Thread Sascha
We're having a problem with calls coming in from our TE110P (an E&M wink T1) through to our queues and then when someone picks up the calls goes dead or silent. They are becoming known as "ghost calls" in our organization. It's seems to only have cropped up in the last couple weeks though we ha

Re: [asterisk-users] Re: [asterisk-biz] Slightly off-topic: Opinions of Comcast and Bellsouth?

2006-08-23 Thread Mike Weaver
I use Speakeasy.net and have been satisfied for a good 4 years now... mogorman wrote: I have used bellsouth dsl and comcast cable. In my experience they both have there problems, but at least in my area I have consistently always gotten anywhere from 2x to 3x more bandwith and reliable rates.

Re: [asterisk-users] Cisco PIX firewall and nat=yes

2006-08-23 Thread Peder @ NetworkOblivion
If you are running a new version of PIX sw (6.3.4 or 6.3.5), then leave fixup on and set "nat=no". The PIX is the only firewall that I have seen that truly does nat correctly. It nat's both the source and dest inside the packet. You can even do reinvite with multiple phones behind a PIX and

[asterisk-users] Connecting Asterisk to Avaya Definity over H.323

2006-08-23 Thread Matt King
Hello, Does anyone out there have experience or settings they can share to help connect Asterisk to an Avaya Definity system over H.323? If so we need your help! Please email me directly. Many thanks, Matt King Managing Director, Orderly Software Ltd. __

RE: [asterisk-users] Compilation

2006-08-23 Thread Dan Austin
The archives should contain these details, but here they are again-   Near line 29- change this line:    app_test.so app_forkcdr.soTo this: app_test.so app_forkcdr.so app_cbmysql.so   Near line 88 add thes lines (just above this line " look:  look.c" app_cbmysql.o: app_cbmys

Re: [asterisk-users] Cisco PIX firewall and nat=yes

2006-08-23 Thread Scott Pinhorne
Hi I use a PIX 515 and had a similar problem when I started. I turned on the fixup for SIP (as well as having nat in sip entry) and it seems to do the trick for me. Good Luck SP Bill Gibbs wrote: Also the phone can dial out from behind the PIX…but obviously not receive calls. Bill

Re: R: [asterisk-users] Snom360 with 6.2.2 firmware

2006-08-23 Thread Brodie Macleod
Well, you could just press the transfer button when the line starts to ring instead of waiting for someone to answer. -Brodie On Wednesday 23 August 2006 02:07 am, Giordano Grandis wrote: > Thanks, but my problem is that I need to transfer a call, while the called > party is ringing. I cannot wa

[asterisk-users] Registering IP Phone To Asterisk

2006-08-23 Thread wyatt . wmvg
What is the process to get an IP phone registered to Asterisk? I bought an Asterisk with a GUI and it has templates for devices such as sipura, cisco, and xten but I am using a Fanvil IP phone. How do I load the template for my IP phone into astrisk so that it can work? Thanks Wyatt __

RE: [asterisk-users] Cisco PIX firewall and nat=yes

2006-08-23 Thread Bill Gibbs
Also the phone can dial out from behind the PIX…but obviously not receive calls.   Bill   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Wednesday, August 23, 2006 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [ast

[asterisk-users] Cisco PIX firewall and nat=yes

2006-08-23 Thread Bill Gibbs
I have a Polycom 501 that works great from behind simple firewalls, like Dlink, etc however behind a Cisco PIX Firewall I see the register messages for the extensions on the Asterisk CLI but when I do a sip show peers I see:   702/702    x.x.x.x D   N  54297    UNR

[asterisk-users] Weird compile problem

2006-08-23 Thread Benjamin Lawetz
I'm in the process of upgrading an asterisk to 1.2.10 and started by upgrading libpri-1.2.3 (make & make install) and zaptel (make & make install). Was about to install asterisk, but doing a "ls" I get the following error: ls: relocation error: /lib/libpthread.so.0: symbol _h_errno, version GLIBC_

Re: [asterisk-users] Trunk with multiple IPs?

2006-08-23 Thread Warren (mailing lists)
In my case it was not a class c, but just 4 separate addresses, one each in NY, Seattle, Miami and London on the Level 3 network. I ended up creating separate entries for each, in and out, and for the outbound route, put all 4 in the order of their ping times. That is working nicely. W _

[asterisk-users] Re: [asterisk-biz] Slightly off-topic: Opinions of Comcast and Bellsouth?

2006-08-23 Thread mogorman
I have used bellsouth dsl and comcast cable. In my experience they both have there problems, but at least in my area I have consistently always gotten anywhere from 2x to 3x more bandwith and reliable rates. but thats just my 2 cents. Mog ___ --Bandwi

Re: [asterisk-users] Snom360 with 6.2.2 firmware

2006-08-23 Thread Andrew Latham
This bridges the call on the phone and not the switch unless I am mistaken On 8/22/06, Brodie Macleod <[EMAIL PROTECTED]> wrote: Although I'm not using this firmware, attended transfers on these phones are done like this (while talking to the person you want to transfer): 1. Press one of

Re: [asterisk-users] Trunk with multiple IPs?

2006-08-23 Thread Rich Adamson
I'm thinking I used deny and permit statements on broadvoice.com way back when, and the configs/sip.conf.sample suggests its still valid for v1.2.10 code. You might take another look at that for sip. Benjamin Lawetz wrote: Agreed that with a other IAX and SIP that have registration informatio

Re: [asterisk-users] Cisco Router QOS and IAX2

2006-08-23 Thread Rich Adamson
Bruce Reeves wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. You can either match on udp/4569, or, match on TOS header bits. I

Re: [asterisk-users] Cisco Router QOS and IAX2

2006-08-23 Thread Dave Fullerton
Bruce Reeves wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. I just spent some time doing this myself. If your routers already

RE: [asterisk-users] Hint extension issue - bug?

2006-08-23 Thread Watkins, Bradley
I may have to eat my words, then. This is the case with trunk, and I can't recall the last time I built a 1.2.x system. I could have sworn that behavior didn't change, but I've been wrong before. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of

Re: [asterisk-users] How to set externip in sip.conf automatically?

2006-08-23 Thread Greg Delgado
The easiest way is to register for free dynamic DNS service at www.dyndns.com. Then use externhost= instead of externip= in sip.conf . If you are using a Linksys router like the WRT54G, it already has a dyndns client which will update the dyndns servers with your ip address everytime it changes.

RE: [asterisk-users] Trunk with multiple IPs?

2006-08-23 Thread Benjamin Lawetz
Agreed that with a other IAX and SIP that have registration information and secrets that works. The problem is when you have a provider that just sends you a SIP call and the only way to identify it is by IP address. In those cases (if I understand correctly) we need a host line don't we? (Or at l

Re: [asterisk-users] Direct to Voicemail

2006-08-23 Thread Aaron Daniel
Since you're using the variables to decide what to do next (VMBOXEXISTSSTATUS), you can go ahead and set priorityjumping=no in the general section of extensions.conf, unless you're using the n+101 priority jumping elsewhere. On Wed, 2006-08-23 at 08:39 -0400, Doug Lytle wrote: > Hey everybody, >

Re: [asterisk-users] Cisco Router QOS and IAX2

2006-08-23 Thread RR
Bruce, this might be able to help give you some hints or a place to start: http://www.voip-info.org/wiki/view/QoS+Cisco Hope that helps \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

RE: [asterisk-users] Hint extension issue - bug?

2006-08-23 Thread Watkins, Bradley
Woah there... Relax, man. I will concur that there are some inconsistencies and things are not exactly how they should be. I'm mostly just pointing out that, for various reasons that I'm not particularly well-equipped to discuss (oej would be able to regale you with the necessary history if you

RE: [asterisk-users] Hint extension issue - bug?

2006-08-23 Thread David Gagnon
On 1.2.10, presence is working very well using friend. The state is refreshing successfully. There is probably antoher problem with your installation cause I'm using hint with friend since 1 years in all my production system. David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL

RE: [asterisk-users] Hint extension issue - bug?

2006-08-23 Thread David Gagnon
Brad, It works with friend. I'm using this config since 1 year. I dunno why it didn't work for Andrew. David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Watkins, Bradley Envoyé : 23 août 2006 08:48 À : Asterisk Users Mailing List - Non-Commercial Di

Re: [asterisk-users] Cisco Router QOS and IAX2

2006-08-23 Thread BJ Weschke
On 8/23/06, Bruce Reeves <[EMAIL PROTECTED]> wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. Can't you just setup a policy class

Re: [asterisk-users] VM - advanced options?

2006-08-23 Thread Doug Lytle
Rich Adamson wrote: running v1.2.10 svn checkout... When I listen to the VM options, it says 'press 3 for advanced options', but after pressing '3', there is nothing there with the exception of pressing '*' to return to the main menu. Rich, If you don't have the dialout option enabled in t

[asterisk-users] Re: [asterisk-biz] Slightly off-topic: Opinions of Comcast and Bellsouth?

2006-08-23 Thread BJ Weschke
On 8/23/06, Alistair Cunningham <[EMAIL PROTECTED]> wrote: Does anyone have an opinion of: 1. Comcast Cable 2. Bellsouth DSL for residential internet and VoIP service? I'm particularly interested in reports on: 1. VoIP voice quality. 2. Any NAT or firewall problems with SIP. 3. How long the

Re: [asterisk-users] Hint extension issue - bug?

2006-08-23 Thread John Novack
Andrew Kohlsmith wrote: This is broken behaviour. I don't know why we have the distinction of users and peers in the first place. A single entry with something along the line of "calltype" taking "incoming", "outgoing" or "both" would be far clearer and eliminate all this inconsistency.

[asterisk-users] Cisco Router QOS and IAX2

2006-08-23 Thread Bruce Reeves
I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. -- BruceNortex Networks ___ --Bandwidth and Colo

[asterisk-users] VM - advanced options?

2006-08-23 Thread Rich Adamson
running v1.2.10 svn checkout... When I listen to the VM options, it says 'press 3 for advanced options', but after pressing '3', there is nothing there with the exception of pressing '*' to return to the main menu. Have I missed a config option, sound file, or is the advanced option not tota

SV: [asterisk-users] Hint extension issue - bug?

2006-08-23 Thread Jon Schøpzinsky
Hello Wouldn't the correct way of handling call limits, be using the Call Group Applications available in Asterisk? Regards Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Andrew Kohlsmith Sendt: 23. august 2006 15:30 Til: asterisk-users@lists.di

RE: [asterisk-users] Trunk with multiple IPs?

2006-08-23 Thread Benjamin Lawetz
Still no answers huh? I've asked a couple of time how to do this, and by the lack of answers, I'm guessing there is no way. The workaround unfortunately is to create an entry for each IP address in the range (I hope you don't have to open up a whole C class) -Original Message- From: [EMA

Re: [asterisk-users] Hint extension issue - bug?

2006-08-23 Thread Andrew Kohlsmith
On Wednesday 23 August 2006 08:48, Watkins, Bradley wrote: > It's not a bug. When you use type=friend, it will create a user object > *and* a peer object. This will make call-limit not function, thereby > breaking hints. There is no reason to use friend anyway. It does not > gain you any functi

Re: [asterisk-users] How to set externip in sip.conf automatically?

2006-08-23 Thread Larry Alkoff
As stated in the original post, when I entter the IP with an editor directly into sip.conf calls work just fine but I am looking for a way to have that done _automatically_. The Asterisk - Future of Telephony book says it is possible for Asterisk to access a Linux environment variable containi

RE: [asterisk-users] PRI and Asterisk

2006-08-23 Thread Kevin Savoy
I have tested Redfone’s boxes. Tried two of them and was able to re-create some issues. I did not have PRI lines but a 24 channel e&m wink line so not sure if PRI is affected as well. I found that over time we had issues with hanging zap channels. Asterisk reported everything was just fine

[asterisk-users] Slightly off-topic: Opinions of Comcast and Bellsouth?

2006-08-23 Thread Alistair Cunningham
Does anyone have an opinion of: 1. Comcast Cable 2. Bellsouth DSL for residential internet and VoIP service? I'm particularly interested in reports on: 1. VoIP voice quality. 2. Any NAT or firewall problems with SIP. 3. How long they take to install the service from date of order. 4. How

Re: [asterisk-users] Strange SIP response

2006-08-23 Thread Joe Dennick
Have you done a "show channels" to see if Asterisk thinks that SIP Device is in use? I experienced this problem once after doing a Blind-Transfer from a Cisco 7940 SIP Phone. The transferred call had long since been disconnected, but the Cisco phone thought it still had control of the call, s

Re: [asterisk-users] Simple CDR parser to print to webpage

2006-08-23 Thread Joe Dennick
Yeah, use Asterisk-Addons and configure the CDR to go into a MySQL database. Once, there, it's really easy to use PHP or Perl to create a custom web-page that shows whatever you want to see. I've got one set up to search for a specific period of time, or for a specific extension. Christopher

Re: [asterisk-users] Hint extension issue - bug?

2006-08-23 Thread Lucas Alvarez
I'm using asterisk 1.2.10 David Gagnon wrote: Are you having this problem with the trunk? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez Envoyé : 22 août 2006 18:23 À : Asterisk Developers Mailing List; Asterisk Users Mailing List - N

RE: [asterisk-users] Missing Extension

2006-08-23 Thread Roger Workman
When I do a sip show peers I see that some phones have lost their registration or is no longer reachable. When this occurs I would like the system to send someone an email that the extension is no longer reachable. Roger Workman Business Development Upperclassman/Universal Holdings LLC Voice: 3

RE: [asterisk-users] Hint extension issue - bug?

2006-08-23 Thread Watkins, Bradley
It's not a bug. When you use type=friend, it will create a user object *and* a peer object. This will make call-limit not function, thereby breaking hints. There is no reason to use friend anyway. It does not gain you any functionality, and in fact breaks some. - Brad -Original Message--

Re: [asterisk-users] No retry after DNS failure

2006-08-23 Thread Andrew Kohlsmith
On Wednesday 23 August 2006 07:07, Remco Barendse wrote: > I am aware that it could mean serious delays for a call to be completed if > the dns lookup was done for every call but surely it should be possible > for * to keep re-trying to resolve an ip address for previous failed > entries let's say

Re: [asterisk-users] Hint extension issue - bug?

2006-08-23 Thread Andrew Kohlsmith
On Wednesday 23 August 2006 06:56, Watkins, Bradley wrote: > This is actually working as designed. You need to use type=peer in order > for call-limit to work properly, which in turn is what allows hints to work > properly. I'm sorry, but type=friend is *SUPPOSED* to equal both user and peer. If

[asterisk-users] Direct to Voicemail

2006-08-23 Thread Doug Lytle
Hey everybody, I've set up an extension that allows users to send a call directly to voice mail. Yesterday, someone accidentally sent a call to an extension that didn't exist and the call was dropped. I found the option to check if a mailbox exists and it works fine, but I get the following

[asterisk-users] Call Handoff

2006-08-23 Thread Scott Pinhorne
Hi All I have 2 phones registered to an asterisk server. The phones are sat behind a NAT. If I have the asterisk sat inline on the call after setting it up (with transfer option specified as an example) the call works fine. If I take out all options so the asterisk should bridge the call an

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