Re: [Asterisk-Users] Sipua SPA-2000 and liong delay after dialling number
You can also adjust the Interdigit Long Timer and Interdigit Short Timer values found in the Regional settings config screen. - Pedro On Fri, 28 Jan 2005 13:36:14 +0100 (CET), Remco Barende <[EMAIL PROTECTED]> wrote: > On Fri, 28 Jan 2005, David John Walsh wrote: > > > The "delay" is a time out. The SPA does not know how many numbers it is > > expecting before it has a complete number for your system. The invite > > message is sent as a single message to asterisk containing the whole number > > string, as apposed to each number individually. > > > > In simple terms you have 2 options at your disposal : > > > > a) encorage users to adopt pressing gate / pound / hash (the noughts and > > crosses board above "9" on the keypad - i cant belive this keyboard doesn't > > have the symbol ;) at the end of the last digit - this in the sipura (like > > 99% of telephony devices) is treated as a send / termination / enter > > instruction and sends the instruction (invite message) to asterisk > > immediatly > > > > Note this only applies if your using a touch-tone / dtmf (dual-tone > > multi-frequency) enabled hand set. > > Great, thanks! This is the easiest solution, the intercom can dial a * and > # I only have to terminate the number with an # :) > > Thanks for the tip! All my visitors at the door will be greatful :) > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipua SPA-2000 and liong delay after diallingnumber
understood - I use the # sign as well, but some users are not used to using the # sign so decreasing the timer helps those that may forget to use the # key. -Pedro On Fri, 28 Jan 2005 08:08:28 -0600, Michael B. Murdock <[EMAIL PROTECTED]> wrote: > Pedro, > > You can also instruct your users to press the # key after dialing the number > to get the dial to start immediately. > > -- Mike > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] outbound 911 calling
You need to create different contexts for each company. - Pedro On Wed, 2 Feb 2005 21:49:53 -0500, Jason Brown <[EMAIL PROTECTED]> wrote: > > > > In order to put a shared pbx in an office building for multiple businesses, > I will have to make sure that the caller ID information going out is > correct. > > > > i.e. company a's main phone number is 5551212 > > > > company b is 5572121 > > > > company c is 5596767 > > > > Now I know how to distribute incoming calls based on the number being > called, but how do you set the caller id going out depending on what company > is dialing out? > > > > if company a dials out, I need to be sure that their correct CID information > is sent out with the call, and ANI if possible. > > > > How does one accomplish this? I tried using the simple fromuser= in > sip.conf, that doesn't work. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: outbound 911 calling
If each company is in their own context, then just specify the callerID for each in their own company-specific outbound context with the SetCallerId command. Since you will have 2 totally different contexts, each company should be isolated to their own set of instructions and thus have 2 different callerid's set. - Pedro On Wed, 2 Feb 2005 22:31:57 -0500, Jason Brown <[EMAIL PROTECTED]> wrote: > > > > Pedro > > > > Exactly my point. I have each company in a different context. How do I > SetCallerID to a number based on the context they are in? > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call forwarding
Cool idea. One question - let's say someone specifies their home phone number and their cell number. How do you take into the account if the cell VM picks up (ie. if cell is out of coverage and VM greeting is played)? On Fri, 04 Feb 2005 10:41:28 -0700, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: > Ryan Courtnage wrote: > > > Can multiple Local channels be safely used in a single Dial command? > > ie: > > > > exten > > => ...,...,Dial(Local/[EMAIL PROTECTED],Local/[EMAIL > > PROTECTED],Local/[EMAIL PROTECTED]) > > Yes, using the standard "&" connector like you would use if you were > dialing multiple SIP peers or any other peers. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using a Dual WAN Load Balancing Device
We have a client that wants to bond 2 DSL circuits instead of getting a T-1 (or similar) at their office to run their VoIP traffic on. We came across this Multihomed Gateway (MH200): http://www.cyberpathinc.com/mh200/details.htm Does anybody think this would work if installed at the client location handling NAT for 10 Cisco 7960's and connecting to our public asterisk server? My concern (as is others on this list in regards to load balancing) is what would happen if a call had to be directed out the other WAN port of the MH200 or if a call were to come in on 1 circuit and it runs out of bandwidth - how would the call be delivered to the second circuit. Or even if during a call, the inbound audio is fine (since DSL usually has more bandwidth on the download), but the outbound audio stream had to be pushed out the other WAN port. Hope that all makes sense (I almost confused myself! LOL) I am not holding my breath that this is a viable solution, but was just wondering your thoughts. Thanks! Pedro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Using a Dual WAN Load Balancing Device
Noah, Thanks for your input on this. I am not sure if it handles incomng connections or not - will have to check. I don't think it will work either - worth a shot to ask though. Thanks! - Pedro On Tue, 8 Feb 2005 10:26:48 -0500, Noah Miller <[EMAIL PROTECTED]> wrote: > > We have a client that wants to bond 2 DSL circuits instead of getting > > a T-1 (or similar) at their office to run their VoIP traffic on. We > > came across this Multihomed Gateway (MH200): > > > > http://www.cyberpathinc.com/mh200/details.htm > > > > Does anybody think this would work if installed at the client location > > handling NAT for 10 Cisco 7960's and connecting to our public asterisk > > server? > > > > My concern (as is others on this list in regards to load balancing) is > > what would happen if a call had to be directed out the other WAN port > > of the MH200 or if a call were to come in on 1 circuit and it runs out > > of bandwidth - how would the call be delivered to the second circuit. > > Or even if during a call, the inbound audio is fine (since DSL usually > > has more bandwidth on the download), but the outbound audio stream had > > to be pushed out the other WAN port. > > > > Hope that all makes sense (I almost confused myself! LOL) > > > > I am not holding my breath that this is a viable solution, but was > > just wondering your thoughts. > > I had the displeasure of working with the now defunct iSurfJanus from > Amplify Networks which is similar to the MH200. I'm not sure the MH200 > is capable of doing what you want it to do. I don't think it does > "incoming load balancing". The only ways I know of to host a machine > behind two or more connections, "incoming load balancing", are 1) > BGP, 2) Cisco HSRP, or with 3) DNS and extremely short TTL values. > There may be some other ways, but these are the popular ones. The > multiple WAN devices capable of incoming load balancing like the F5 > BigIP, Fatpipe Products, Radware Linkproof, etc. all use special DNS > entries to spread the incoming connections between WAN connections. > > When I looked at the product specs of the MH200 it makes no mention of > BGP, DNS, or anything else that might handle incoming connections. In > fact, it doesn't say anything about incoming connections at all. > > To answer your question directly, I don't know how the other products > work, but I could configure the iSurfJanus to respond to requests only > on the same connection they came in on. If the MH200 does handle > incoming connections, you will probably need to ask the folks that make > it if you can explicitly specify to respond to incoming request on the > same WAN connection they came in on. > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zombie SIP channels
Does anyone know how to kill a zombie channel? Here is what I see on a show channels: -- show channels Channel (ContextExtensionPri ) State Appl. Data SIP/frontdesk-72c7 (customercontext 1 ) Up Bridged Call SIP/frontdesk-0461 SIP/frontdesk-0461 (customercontext 100 1 ) Ring Dial SIP/frontdesk|20|t 2 active channel(s) -- No one is on a call - how can I get rid of this without restarting asterisk? Thanks! Pedro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zombie SIP channels
I tried to send this earlier but does not look like it went through for some reason. If you get this twice - my appologies. Does anyone know how to kill a zombie channel (and why do they pop up)? Here is what I see on a show channels: -- show channels Channel (ContextExtensionPri ) State Appl. Data SIP/frontdesk-72c7 (customercontext 1 ) Up Bridged Call SIP/frontdesk-0461 SIP/frontdesk-0461 (customercontext 100 1 ) Ring Dial SIP/frontdesk|20|t 2 active channel(s) -- No one is on a call - how can I get rid of this without restarting asterisk? Thanks! Pedro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zombie SIP channels
Thanks for the tip. They both seemed to go away on their own after a while with no action on my part. I am not sure what caused it (there is nothing in the log file). This is the first time I have seen it on any of my asterisk machines (and I have been working with asterisk for a year now). Any ideas on why a zombie sip channel would occur? Thanks in advance for any insight on this. - Pedro On Thu, 10 Feb 2005 14:57:17 +1300, Matt Riddell <[EMAIL PROTECTED]> wrote: > Pedro wrote: > > No one is on a call - how can I get rid of this without restarting asterisk? > > soft hangup in Asterisk console. > > It'd pay to try and find out why you're getting them though. > > :) > > -- > Cheers, > > Matt Riddell > ___ > > http://www.sineapps.com/news.php (Daily Asterisk News - html) > http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zombie SIP channels
Thanks for the feedback! Running CVS-v1-0-11/12/04 (stable) on Fedora Core 1 with Cisco 7960G's. Asterisk server is on public IP and Cisco 7960G is at client location NAT-ed behind a Cisco soho91-k9 with nine other Cisco 7960G's (each phone has registration expiring every 120 seconds). Here is excerpt from sip.conf [general] disallow=all allow=ulaw port=5060 context=incoming maxexpirey=3600 defaultexpirey=300 canreinvite=no tos=reliability srvlookup=yes videosupport=no dtmfmode=inband nat=yes insecure=very [frontdesk] context=customer type=friend username=frontdesk secret=password host=dynamic canreinvite=no [EMAIL PROTECTED] nat=yes qualify=yes callerid="Front Desk" <100> accountcode=customer amaflags=billing This is the first time I have seen this so it does not appear to happen too often. Obviously would rather not upgrade if possible has everything seems running fine. But good to know that if it becomes a problem, I can try upgrading to 1.0.3 or later. Thanks! Pedro On Thu, 10 Feb 2005 09:19:45 +0100, Florian Overkamp <[EMAIL PROTECTED]> wrote: > Hi, > > > -Original Message- > > Does anyone know how to kill a zombie channel? > > > > Here is what I see on a show channels: > > -- > > show channels > > Channel (ContextExtensionPri ) State Appl. > > Data > > SIP/frontdesk-72c7 (customercontext 1 ) Up > > Bridged Call SIP/frontdesk-0461 > > SIP/frontdesk-0461 (customercontext 100 1 ) > > Ring Dial SIP/frontdesk|20|t > > 2 active channel(s) > > -- > > > > No one is on a call - how can I get rid of this without > > restarting asterisk? > > This was an issue in older versions of asterisk. It would help if you could > tell us what setup you are running. > If this is infact your problem too, a simple update of your asterisk to > 1.0.3 or later will help. > > Florian > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zombie SIP channels
What is odd is no meetme is being used. But may be related - thanks! Pedro On Thu, 10 Feb 2005 14:37:31 +0100, Florian Overkamp <[EMAIL PROTECTED]> wrote: > Hi, > > > -Original Message- > > This is the first time I have seen this so it does not appear to > > happen too often. Obviously would rather not upgrade if possible has > > everything seems running fine. But good to know that if it becomes a > > problem, I can try upgrading to 1.0.3 or later. > > If my memory serves me correctly, this is the issue: > > http://bugs.digium.com/bug_view_page.php?bug_id=0002938 > > It's a two line fix, so if you want you can easily verify and apply manually > so you don't have to introduce any other new code. > > Florian > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zombie SIP channels
Ok this is odd - caught it again twice today. The more I thought about what has changed on the server I realized that I was not using a timing device before, but am now using ztdummy. I if that could be causing the zombies? - Pedro On Thu, 10 Feb 2005 08:50:35 -0500, Pedro <[EMAIL PROTECTED]> wrote: > What is odd is no meetme is being used. But may be related - thanks! > > Pedro > > > On Thu, 10 Feb 2005 14:37:31 +0100, Florian Overkamp > <[EMAIL PROTECTED]> wrote: > > Hi, > > > > > -Original Message- > > > This is the first time I have seen this so it does not appear to > > > happen too often. Obviously would rather not upgrade if possible has > > > everything seems running fine. But good to know that if it becomes a > > > problem, I can try upgrading to 1.0.3 or later. > > > > If my memory serves me correctly, this is the issue: > > > > http://bugs.digium.com/bug_view_page.php?bug_id=0002938 > > > > It's a two line fix, so if you want you can easily verify and apply manually > > so you don't have to introduce any other new code. > > > > Florian > > > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura g729 call quality to PSTN
If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side (the audio on the phone connected to the Sipura sounds fine). My guess is that the Sipura does not compress the outbound audio very effectively and since the incoming audio from the PSTN is already compressed by the VoIP provider, it is just delivering the good-sounding g729 stream. It is worth noting that call quality on both the IP and PSTN side is great when using the Cisco 7960 with g729. It is just with the Sipura that the sound quality on the PSTN-side sounds like a bad quality cell phone call. I even got an SPA-2100 in hopes that the g729 would sound better on that unit, but the same issue is present there as well. Is it just a bad implementation of g729 compression with the Sipura product line? Any thoughts or recommendations are appreciated :) Thanks! - Pedro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip phones how to dial a # sign?
I have had this same problem. The only way I know is to disable transfers in asterisk. You can still use the transfer control in your SIP device. Of course this does not work with call parking. I would be very interested in a solution that does not require disabling of transfers in asterisk as well. Pedro On Tue, 15 Feb 2005 09:52:56 +0100 (CET), Remco Barende <[EMAIL PROTECTED]> wrote: > Hi list! > > I have some sip phones and Sipura ATA 2000's. However after dialling a > number I need to dial a # to control a device. > > When I dial # Asterisk kicks in and puts the call on hold. How can I > change this? > > Thx!! > Remco > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sixtel.net / IAX.CC - Vanity Toll-Free Number
Same boat here. Actually got someone on AOL instant messenger yesterday. Their response as follows when asked how long it will take to get our 800 number: [15:11] sixtel9: it's in the works any time frame? [15:14] sixtel9: not specifically, we switched carriers so we're dealing w/ some issues just need to know if it will be weeks/months/ or days [15:21] sixtel9: days - Pedro On Tue, 15 Feb 2005 09:41:12 -0500 (EST), Paul Dugas <[EMAIL PROTECTED]> wrote: > On Tue, February 15, 2005 9:27 am, Rob Risner said: > > I'm just wondering, how long should a vanity number transfer really take? > > No help here, just posting a "me too" to warn others. Friday was 10 days > for me. No happy to hear you've waited much longer with the same result. > Can never raise them on the phone. They take days to respond to the > ticket and are rather terse when they actually do. > > Not pleased at all. > > Paul > > -- > Paul A. DugasDugas Enterprises, LLC > [EMAIL PROTECTED]1711 Indian Ridge Drive > p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip phones how to dial a # sign?
Is there a way to somehow do an "escape" # so that you can still use the # key to control devices that require a #, but still keep the T in the dial plan? We have clients that need to check external voicemail systems that require the use of the # sign, but still want to have the call parking feature. On Tue, 15 Feb 2005 06:54:23 -0700, Michael Welter <[EMAIL PROTECTED]> wrote: > Remco Barende wrote: > > Hi list! > > > > I have some sip phones and Sipura ATA 2000's. However after dialling a > > number I need to dial a # to control a device. > > > > When I dial # Asterisk kicks in and puts the call on hold. How can I > > change this? > > Do you have the "T" in your Dial statment? Remove the "T" and try it. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
Couple of days. Apparently the new US carrier has some changes that needs to be made. On 6/14/05, Wiley Siler <[EMAIL PROTECTED]> wrote: > Did they say when it would be corrected? > > W > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Pedro > Sent: Tuesday, June 14, 2005 9:22 AM > To: Matt > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Anyone noticed Voipjet voice quality > problems? > > Caller ID is still not working to certain areas. This problem was > confirmed by voipjet tech support in their last e-mail to me. > > On 6/13/05, Matt <[EMAIL PROTECTED]> wrote: > > I never noticed any problems.. so I can't comment :) hehe > > > > On 6/11/05, Pedro <[EMAIL PROTECTED]> wrote: > > > Finally got a response from voipjet support and they say they have > > > switched to a new provider for US termination. I have yet to test > > > this out as I have not had a chance to build them back into our > > > routes but will report my findings once I do. Anyone else notice > > > any improvements? > > > > > > On 6/9/05, Moody <[EMAIL PROTECTED]> wrote: > > > > We have been having serious quality problems using the westcoast > > > > server - been using the East coast server with increased success > > > > but seeing some issues related to going cross continent. > > > > > > > > Voipjet, you listening? > > > > ___ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@lists.digium.com > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > To UNSUBSCRIBE or update options visit: > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
Looks like 9 out of 10 calls are failing on voipjet at the moment (at least terminating to South Florida numbers). Keep getting message that says "number can not be completed as dialed". Anyone else seeing this? On 6/15/05, Pedro <[EMAIL PROTECTED]> wrote: > Couple of days. Apparently the new US carrier has some changes that > needs to be made. > > On 6/14/05, Wiley Siler <[EMAIL PROTECTED]> wrote: > > Did they say when it would be corrected? > > > > W > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Pedro > > Sent: Tuesday, June 14, 2005 9:22 AM > > To: Matt > > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Anyone noticed Voipjet voice quality > > problems? > > > > Caller ID is still not working to certain areas. This problem was > > confirmed by voipjet tech support in their last e-mail to me. > > > > On 6/13/05, Matt <[EMAIL PROTECTED]> wrote: > > > I never noticed any problems.. so I can't comment :) hehe > > > > > > On 6/11/05, Pedro <[EMAIL PROTECTED]> wrote: > > > > Finally got a response from voipjet support and they say they have > > > > switched to a new provider for US termination. I have yet to test > > > > this out as I have not had a chance to build them back into our > > > > routes but will report my findings once I do. Anyone else notice > > > > any improvements? > > > > > > > > On 6/9/05, Moody <[EMAIL PROTECTED]> wrote: > > > > > We have been having serious quality problems using the westcoast > > > > > server - been using the East coast server with increased success > > > > > but seeing some issues related to going cross continent. > > > > > > > > > > Voipjet, you listening? > > > > > ___ > > > > > Asterisk-Users mailing list > > > > > Asterisk-Users@lists.digium.com > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > To UNSUBSCRIBE or update options visit: > > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@lists.digium.com > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > To UNSUBSCRIBE or update options visit: > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to send voicemail notifcation every 15 minutes until message is checked
I have searched quite a few places and have not seen this discussed. Basically I was wondering how would you go about having an option for a user to be notified every 15 minutes until their new voicemail message is checked. Since the notification e-mails we send get sent to cell phones or actual pagers (via e-mail), there are times when a person is out of range and misses a page or just simply is too busy to check voicemail and then forgets. They want to be reminded 15 minutes later until that new message is checked. Current version of asterisk that we are running is CVS-v1-0-11/12/04 (which has been running rock-solid I might add). Any thoughts are appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to send voicemail notifcation every 15 minutes until message is checked
Thanks - a cronjob for the user was going to be my last resort. Was not sure if there was a setting like "repeatnotify=15" to repeat the notice every 15 minutes. Thanks for your feedback though! On 7/1/05, Michiel van Baak <[EMAIL PROTECTED]> wrote: > On 13:33, Fri 01 Jul 05, Pedro wrote: > > I have searched quite a few places and have not seen this discussed. > > Basically I was wondering how would you go about having an option for > > a user to be notified every 15 minutes until their new voicemail > > message is checked. Since the notification e-mails we send get sent > > to cell phones or actual pagers (via e-mail), there are times when a > > person is out of range and misses a page or just simply is too busy to > > check voicemail and then forgets. They want to be reminded 15 minutes > > later until that new message is checked. > > > > Current version of asterisk that we are running is CVS-v1-0-11/12/04 > > (which has been running rock-solid I might add). Any thoughts are > > appreciated. > > Hi, > > You can check the new mail count with the manager interface > or by looking at the spool dir. > If you put this in cron every 15 minutes, you're done. > > Michiel > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Logging SIP response codes
Is there a way to log SIP response codes without enabling verbose logging? Reason being is that from time to time I see a call fail on our primary provider and roll-over to our backup providers. If I happen to catch it on the console I can see the code "484" or similar. It would really help in troubleshooting with our primary provider if I could log those types of codes. Verbose just saves way to much stuff in the log files. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Logging SIP response codes
Had not seen a response on the following question - wondering if anyone may have any insight on this? Original Question- Is there a way to log SIP response codes without enabling verbose logging? Reason being is that from time to time I see a call fail on our primary provider and roll-over to our backup providers. If I happen to catch it on the console I can see the code "484" or similar. It would really help in troubleshooting with our primary provider if I could log those types of codes. Verbose just saves way to much stuff in the log files. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mitel Ip phone ?
Will let you know - getting one soon to test. On Feb 13, 2005 10:58 PM, eric m <[EMAIL PROTECTED]> wrote: > Hi, > > I really appreciate the look and design of newer Mitel Ip phone. > > I search througt the list and found only fews notes about the use Mitel 5055 > phone on *. Anyone use other model (especially 52xx series) on * ?? > Compatible? Easy to use? hassle to configure? > > Thansk for your suggestion! > > Best Regards, > > eric. > [EMAIL PROTECTED] > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF intermittently stops working
Every so often I get a report from a customer that DTMF stops working while checking voicemail. The customer has to hang up and check for messages again. I have actually had this happen to me twice in the past 6 months so I know it does happen, just not very often. So far, the only incidents have been with Cisco 7960's. I was just wondering if anyone had noticed this behavior in their environment. We are using ulaw and rfc2833 with the following configuration (Asterisk CVS-v1-0-11/12/04): SIP Provider (SIP)(SIP) Asterisk Gateway (IAX)(IAX) Customer Asterisk Server (SIP)(SIP) Cisco 7960 Any thoughts are appreciated. Thank You, Pedro TRACI.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone problems to non-na numbers
Yes, same problem here. Sign-ed up with VoipJet and seems to work just fine (prices for most areas we call are cheaper too from what I saw). Only been using them for 24 hours so can't say much about long-term stability, but so far so good. Pedro On 4/19/05, Matthew Asham <[EMAIL PROTECTED]> wrote: > Is anyone else having problems with Nufone dialing international (non > NA) numbers? > > Pretty much every intl number dialed comes up with a voice intercept > saying the call could not be completed as dialed. Tried it with two > separate accounts, and the numbers themselves work from the local > telco. > > The problem appears to have started within the last few days (and yes I > have emailed [EMAIL PROTECTED], just wondering if we're the only ones > having the problem). > > Matthew > > -- > Matthew Asham - the B.C. Wireless Network Society > www.bcwireless.net - +1 604 484 5289 x1006 > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mitel Ip phone ?
Just got the new Mitel 5220 Dual Mode IP Phone and I have to say I am impressed so far. Changing to SIP mode is VERY easy as long as you have the SIP firmware which can be downloaded by going to: http://sipdnld.mitel.com Phone audio quality is excellent. Look and feel are also good. The phone can only accept 1 SIP user account, but can handle 4 simultaneous calls. Conferencing can be done with 2 of the 4 simultaneous calls and you can switch between any of the calls at any time. The only issue I see is the small display. Clearing a missed call involves cycling through a few menus to clear the missed call log (not sure there is a short cut for this). The speed dial buttons fine, but you must manually write the name of each speed dial button (Cisco-type LCD would be nice for this but would probably add to the cost of the phone). Well, that is my initial impressions. If I come across anything else important I will let you know. Pedro TRACI.net On 4/4/05, Kris Edwards <[EMAIL PROTECTED]> wrote: > Here's a good sign: > > Mitel is also addressing economy in adding SIP compliance to two of its > IP phones. The Mitel 5215 and 5220 run Mitel's proprietary MiNET > protocol for operation with the ICP 330 but also will run SIP, allowing > users to point them at such SIP-based PBXes as Asterisk's or Snom's, or > to another SIP proxy server. > > _ > > I know Mitel's older models can be changed to sip (there's a howto on > voip-info) so surely there is hope for the newer models. > > Here's the article the above is taken from: > > http://www.thechannelinsider.com/article2/0,1759,1725518,00.asp > > > Kris > > > Pedro wrote: > > Will let you know - getting one soon to test. > > > > On Feb 13, 2005 10:58 PM, eric m <[EMAIL PROTECTED]> wrote: > > > >>Hi, > >> > >>I really appreciate the look and design of newer Mitel Ip phone. > >> > >>I search througt the list and found only fews notes about the use Mitel 5055 > >>phone on *. Anyone use other model (especially 52xx series) on * ?? > >>Compatible? Easy to use? hassle to configure? > >> > >>Thansk for your suggestion! > >> > >>Best Regards, > >> > >>eric. > >>[EMAIL PROTECTED] > >> > >>___ > >>Asterisk-Users mailing list > >>Asterisk-Users@lists.digium.com > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >>To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
Have you tried to enable NAT translation on the Grandstream? On 4/23/05, Tomas Florian <[EMAIL PROTECTED]> wrote: > I'm trying to register BT100s ... (doesn't work) > X-Lite seems to work though > > Tomas > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo > Sent: Saturday, April 23, 2005 8:48 PM > To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G > > Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running > behind my Linksys WTR43GS with no issues. This is at home registering to an > external * box and to vonage. > > - Original Message - > From: "Luki" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Saturday, April 23, 2005 9:41 PM > Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G > > The WRT54G work fine... > > I have a Sipura 1000 and a Grandstream 286, both nated through a > WRT54G on a single public IP. Worked "out of the box" -- no special > settings needed. I was even surprised that I did not need to turn on > the NAT handling in the Sipura ATA. > > Then I have a WRT54G running as a wireless client, and a Sipura 1001 > connected to it, essentially behind two NAT's. Works fine too. > > --Luki > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF intermittently stops working
Thank you for your feedback. I was mearly wondering if others had experienced this issue in "their" environments. Was not trying to open a bug report or officially report an issue. Strictly a curiousity request. Really do not want to upgrade if everything else works fine. Since this issue happens so intermittently, I would have no way of testing if the new version would fix it since I could go for 6 months without having the issue on my current version (no way to consistently replicate the problem). If you have a way to consistently replicate this issue, I would appreciate that information. I can assure you I exhausted search options and researched this issue elsewhere with little success before posting my question here to avoid wasting people's time. On 4/24/05, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: > Joseph wrote: > > > We have the same problem with 7960, just randomly it will stop *hearing* > > the dtmf tones and you have to hangup and call back. > > This problem was fixed in CVS long ago, and current stable releases have > the fix as well. When you are running a copy of Asterisk that is 4/5 > months old, it's better to update first before reporting a problem, > since it may already have been fixed. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mitel Ip phone ?
Just wanted to correct this last post - apparently, you can configure the other speed buttons to also be separate lines with their own SIP account. On 4/22/05, Pedro <[EMAIL PROTECTED]> wrote: > Just got the new Mitel 5220 Dual Mode IP Phone and I have to say I am > impressed so far. Changing to SIP mode is VERY easy as long as you > have the SIP firmware which can be downloaded by going to: > > http://sipdnld.mitel.com > > Phone audio quality is excellent. Look and feel are also good. The > phone can only accept 1 SIP user account, but can handle 4 > simultaneous calls. Conferencing can be done with 2 of the 4 > simultaneous calls and you can switch between any of the calls at any > time. The only issue I see is the small display. Clearing a missed > call involves cycling through a few menus to clear the missed call log > (not sure there is a short cut for this). The speed dial buttons > fine, but you must manually write the name of each speed dial button > (Cisco-type LCD would be nice for this but would probably add to the > cost of the phone). > > Well, that is my initial impressions. If I come across anything else > important I will let you know. > > Pedro > TRACI.net > > On 4/4/05, Kris Edwards <[EMAIL PROTECTED]> wrote: > > Here's a good sign: > > > > Mitel is also addressing economy in adding SIP compliance to two of its > > IP phones. The Mitel 5215 and 5220 run Mitel's proprietary MiNET > > protocol for operation with the ICP 330 but also will run SIP, allowing > > users to point them at such SIP-based PBXes as Asterisk's or Snom's, or > > to another SIP proxy server. > > > > _ > > > > I know Mitel's older models can be changed to sip (there's a howto on > > voip-info) so surely there is hope for the newer models. > > > > Here's the article the above is taken from: > > > > http://www.thechannelinsider.com/article2/0,1759,1725518,00.asp > > > > > > Kris > > > > > > Pedro wrote: > > > Will let you know - getting one soon to test. > > > > > > On Feb 13, 2005 10:58 PM, eric m <[EMAIL PROTECTED]> wrote: > > > > > >>Hi, > > >> > > >>I really appreciate the look and design of newer Mitel Ip phone. > > >> > > >>I search througt the list and found only fews notes about the use Mitel > > >>5055 > > >>phone on *. Anyone use other model (especially 52xx series) on * ?? > > >>Compatible? Easy to use? hassle to configure? > > >> > > >>Thansk for your suggestion! > > >> > > >>Best Regards, > > >> > > >>eric. > > >>[EMAIL PROTECTED] > > >> > > >>___ > > >>Asterisk-Users mailing list > > >>Asterisk-Users@lists.digium.com > > >>http://lists.digium.com/mailman/listinfo/asterisk-users > > >>To UNSUBSCRIBE or update options visit: > > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > >> > > > > > > ___ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Confused on G723 and G729
In your case, where you will need the license is on the box that your phones register to. For exampe, when someone checks voicemail, encoding takes place, therefore you need a license. Look at it this way: [g729 provider] -(SIP or IAX)--- [g729 asterisk server] - no license required in the above connection if using g729 solely [g729 asterisk server containing non-g729 audio files] (SIP)- [g729 SIP Phone] - a license is required above for each non-g729 audio file or stream that needs to be encoded to be sent out as g729 to the g729 SIP Phone (ie. voicemail, IVR prompts, etc.). Hope that makes sense. On 4/28/05, Matt <[EMAIL PROTECTED]> wrote: > For instance.. when I try to use G723.1 on my phone (and just call in > from my PRI line) I get: > Unable to find a path from g723 to ulaw. > Unable to find a path from ulaw to g723. > No path to translate from Zap/1-1(68) to Sip/201-80c7(1). > Same things happens if I call in on my current provider's number which > uses G711 for the codec. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 license
Actually called Digium with this exact question last week. They said that you can register the new license on the new server provided that you ony registered it once before. They said there is no "unregister" script to unregister the license from the old server, however. If you have already used up your 2 registrations, you will need to contact Digium for assistance on this. I also asked if leaving the keys on my dev. box would cause a conflict (also was pretty clear that I wanted to be in compliance with their license agreement) and the lady said there was no problem and leaving the old keys on the dev. box would not cause a conflict. On 5/2/05, Peter <[EMAIL PROTECTED]> wrote: > Hi all. > > Dopes someone know how I can move a key license of the g729 > codec from one to another machine? > Find nothing usefull @ the wiki. > > Thnx 4 help in advance. > > Regards. > > -Peter > > -- > Please no HTML, I'm not a browser > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A good SIP receptionist phone
What I did once was create an announcement that got played to the receptionist announcing who the call was for based on the number that was called. This allowed the receptionist to know which greeting to recite. On 5/2/05, Michael Welter <[EMAIL PROTECTED]> wrote: > Chris Mason (Lists) wrote: > > The user name is the extension and the password is always the same. Not hard > > to configure. > > > With the SNOM 220, you have five buttons/lamps that can be used as > "line" appearances--these buttons can each register to a different SIP URL. > > Each sidecar has 20 buttons/lamps, and you may have up to three > sidecars. Using the "hint" priority in Asterisk, the buttons serve as > extension busy lamps. You can also use these buttons to transfer calls. > > I have an executive suites customer where each tenant is a separate > business. For an incoming call, the attendant needs to know which DID > number is being called so she can answer with the proper greeting. > > I would like the sidecar buttons to be able to register to a SIP URL, so > an incoming call would blink the tenants button, but that is not > possible--I can only use the five buttons on the phone for that purpose, > and there are more than five tenants. > > A suggestion was to alter the Called ID Name to the DID number. This > would work for the attendant, but the tenant would like to see the > original Caller ID Name. > > I would rather not have to put a PC at the attendants position, but that > is the way this is shaping up. Does anyone have any suggestions? > > Thanks, > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FCC Will Force VOIP E911 in 120 days ?
http://www.lightreading.com/document.asp?doc_id=73943&site=lightreading I know e911 has been discussed on ths list before, but I just read this and it got me thinking that if you have a Wholesale VoIP carrier - wouldn't they have to pass e911 on to you as a VoIP provider to, in turn, pass on to your end-users? Of course there would be a fee - just wondering if this is how the "start-ups" will be able to reach a deadline on this if it passes. Especially since it seems you have to be a CLEC to interface with the PASP database from the threads I have been reading. Also, how in the world will this work with hosted IP-PBX solutions where the customer may have their employees scattered around the country working from their homes? Since all their outbound calls share the same callerID which may not even be in the local area that they are physically located in, how will the call get routed to the proper PASP (and better yet - how will the PASP know which employee called them)? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to detect DTMF and change if needed
I have done some searching and not sure this is even possible, but here it goes... **Scenario** Let's say you have an asterisk server that you use to connect to a SIP provider that you push your PSTN-bound calls to using g711 and out-of-band DTMF. The SIP phones in use are Cisco 7960's and are set to also use out-of-band DTMF. For the most part, everything works great. However, a few numbers that are dialed and pushed to the SIP provider that get connected to a remote IVR system seem to have DTMF issues where no digits are recognized. A call to the SIP provider confirms that certain calls get routed to one carrier while others get routed to other carriers and the numbers that are showing the DTMF issues are the carriers that they peer with that do not support out-of-band DTMF with the g711 codec. When asked if they could translate our out-of-band DTMF signals to a compatible format that their carrier requires, they bascally say that while that is possible, they will not do it. **The Question** So here is my question - is it possible to detect the DTMF mode of the call and if out-of-band is not supported, can you change it to inband as a last resort? Is there a way to set priority for DTMF signalling like you can do with codecs? I have tried that (see below) but it seems to default to inband (is this even a proper way to handle 2 DTMF modes?). [sipprovider] type=friend host=xxx.xxx.xxx.xxx disallow=all allow=ulaw maxexpirey=15 dtmfmode=rfc2833 dtmfmode=inband nat=no insecure=very canreinvite=no I have searched and searched and the closest thing that I have found is "SIPDtmfMode" but from what it looks like it needs to be initiated before the call is placed. By the way - the reason inband is not being used is that digit accuracy is terrible with the inband setting. Any thoughts are appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
Definately problems with voice quality and caller ID is not working very well. I have e-mail a couple times and still no response from their tech support on this. This is very concerning since I tried all 3 servers with the same results. On 6/8/05, Julio Arruda <[EMAIL PROTECTED]> wrote: > Roman Zhovtulya wrote: > > Dear all, > > I've noticed some significant voice quality deterioration when calling US > > landline via VoIPjet.com in the last week or so. > > Before that the quality was pretty good. > > Has anyone else experienced any voice quality problems with voipjet > > recently? > > I've been using VOIPJET for Brazil LD without any problems. > (or should I say, my wife has been using, still can't thank VOIP enough > for the savings..) > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
Seems things have just got worse. Just got reports that 800 numbers are not terminating. For example, can not dial: 800-888-9358 or 800-922-4684 Had to pull voipjet out of our routes until this gets fixed. On 6/9/05, Moody <[EMAIL PROTECTED]> wrote: > We have been having serious quality problems using the westcoast > server - been using the East coast server with increased success but > seeing some issues related to going cross continent. > > Voipjet, you listening? > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
We are a VoIP provider and need to push out 100,000 - 200,000 minutes per month (ie. need a carrier-level package - not a Vonage, etc.). To date I have not found a wholesale SIP/IAX VoIP provider provide 800 termination for free. However, if you have one, please provide the information and I will definately check them out. On 6/10/05, Pedro <[EMAIL PROTECTED]> wrote: > Please provide the SIP or IAX provider you are using that allows you > to terminate to 800 numbers for free. > > On 6/10/05, Matt <[EMAIL PROTECTED]> wrote: > > Why would you even be routing 800 numbers out voipjet? They CHARGE you! > > > > On 6/10/05, Pedro <[EMAIL PROTECTED]> wrote: > > > Seems things have just got worse. Just got reports that 800 numbers > > > are not terminating. For example, can not dial: > > > > > > 800-888-9358 > > > or > > > 800-922-4684 > > > > > > Had to pull voipjet out of our routes until this gets fixed. > > > > > > On 6/9/05, Moody <[EMAIL PROTECTED]> wrote: > > > > We have been having serious quality problems using the westcoast > > > > server - been using the East coast server with increased success but > > > > seeing some issues related to going cross continent. > > > > > > > > Voipjet, you listening? > > > > ___ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@lists.digium.com > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > To UNSUBSCRIBE or update options visit: > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
Finally got a response from voipjet support and they say they have switched to a new provider for US termination. I have yet to test this out as I have not had a chance to build them back into our routes but will report my findings once I do. Anyone else notice any improvements? On 6/9/05, Moody <[EMAIL PROTECTED]> wrote: > We have been having serious quality problems using the westcoast > server - been using the East coast server with increased success but > seeing some issues related to going cross continent. > > Voipjet, you listening? > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
Caller ID is still not working to certain areas. This problem was confirmed by voipjet tech support in their last e-mail to me. On 6/13/05, Matt <[EMAIL PROTECTED]> wrote: > I never noticed any problems.. so I can't comment :) hehe > > On 6/11/05, Pedro <[EMAIL PROTECTED]> wrote: > > Finally got a response from voipjet support and they say they have > > switched to a new provider for US termination. I have yet to test > > this out as I have not had a chance to build them back into our routes > > but will report my findings once I do. Anyone else notice any > > improvements? > > > > On 6/9/05, Moody <[EMAIL PROTECTED]> wrote: > > > We have been having serious quality problems using the westcoast > > > server - been using the East coast server with increased success but > > > seeing some issues related to going cross continent. > > > > > > Voipjet, you listening? > > > ___ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura g729 call quality to PSTN
uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler <[EMAIL PROTECTED]> wrote: > > On Feb 14, 2005, at 1:25 PM, Pedro wrote: > > > > > Is it just a bad implementation of g729 compression with the Sipura > > product line? > > > > That would be my guess. > > -mark > > -- > Mark Eissler, [EMAIL PROTECTED] > Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip phones how to dial a # sign?
> Your SIP device does not support attended transfers? Yes they do > If your devices support their own transfer feature (odd enough usually > labeled "Transfer") > then there is NO REASON to use T/t transfers. Call parking can only work with T/t transfers (at least on the version I am running - CVS Stable 11/12/2004) > If your SIP devices do not support their own transfer > option then either you didn't do enough research before you installed > Asterisk or you were just too cheap when buying phones. obviously not the case here (Cisco 7960's are not cheap and a lot of research was done) > Do you really need to park outbound external calls? I don't, but our customers have interesting needs :) Actually even if you only enable the t transfer and disable the T transfer, one of our users still has an issue: Scenario: The user's job is to train a customer on how to use an Octel voicemail system. We will call our user UserA and the customer who is to be trained UserB. When UserB calls in from the PSTN to UserA, UserA creates a 3-way call between UserB and the Octel voicemail system. In order to use the features of the Octel voice mail system, you must use a # key. When UserA presses the # key to use the Octel system, UserB is placed on hold ready for transfer. Of course the easy solution would be to have UserA hang up and call UserB back and place the 3-way call that way (since T transfers are disabled), however, when presented with that option, UserA did not approve of the solution. For now we have had to disable the t transfer (and call parking as well and rely on the SIP device's attended transfer) until we can figure out how to work around this issue. Someone made a comment about features.conf in a later version which I will have to investigate as well. Thanks, Pedro On Tue, 15 Feb 2005 11:35:12 -0600, Eric Wieling <[EMAIL PROTECTED]> wrote: > Pedro wrote: > > > Is there a way to somehow do an "escape" # so that you can still use > > the # key to control devices that require a #, but still keep the T in > > the dial plan? We have clients that need to check external voicemail > > systems that require the use of the # sign, but still want to have the > > call parking feature. > > Your SIP device does not support attended transfers? That really > sucks. T and t are cool hacks for devices that do not support > transfers. If your devices support their own transfer feature (odd > enough usually labeled "Transfer") then there is NO REASON to use T/t > transfers. If your SIP devices do not support their own transfer > option then either you didn't do enough research before you installed > Asterisk or you were just too cheap when buying phones. > > Do you really need to park outbound external calls? > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura g729 call quality to PSTN
Actually the SPA-2100 supports 2 g729 channels which is why I bought it. Unfortunately, the call quality is just as poor on the 2100 as it is on the 2000. - Pedro On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan <[EMAIL PROTECTED]> wrote: > Is it just a bad implementation of g729 compression with the Sipura > > > > product line? > > > > > > > > That would be my guess too . why SPA-2000 supports G729 for one > channel only? no enough CPU power to code/decode G.729 for two > channels? > > Jeffey > > www.mutualphone.com > > > On Tue, 15 Feb 2005 16:31:59 -0500, Pedro <[EMAIL PROTECTED]> wrote: > > uggg. > > > > Is anyone out there having any luck with the SPA-2000 or SPA-2100 > > using the g729 codec with decent call quality? > > > > > > On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler <[EMAIL PROTECTED]> wrote: > > > > > > On Feb 14, 2005, at 1:25 PM, Pedro wrote: > > > > > > > > > > > Is it just a bad implementation of g729 compression with the Sipura > > > > product line? > > > > > > > > > > That would be my guess. > > > > > > -mark > > > > > > -- > > > Mark Eissler, [EMAIL PROTECTED] > > > Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com > > > > > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip phones how to dial a # sign?
FYI: Found the info on the wiki regarding features.conf: http://voip-info.org/tiki-index.php?page=Asterisk%20config%20features.conf On Tue, 15 Feb 2005 13:10:40 -0500, C F <[EMAIL PROTECTED]> wrote: > Use the latest stable or CVS HEAD and modify features.conf. You can > change it there. > > > On Tue, 15 Feb 2005 11:35:12 -0600, Eric Wieling <[EMAIL PROTECTED]> wrote: > > Pedro wrote: > > > > > Is there a way to somehow do an "escape" # so that you can still use > > > the # key to control devices that require a #, but still keep the T in > > > the dial plan? We have clients that need to check external voicemail > > > systems that require the use of the # sign, but still want to have the > > > call parking feature. > > > > Your SIP device does not support attended transfers? That really > > sucks. T and t are cool hacks for devices that do not support > > transfers. If your devices support their own transfer feature (odd > > enough usually labeled "Transfer") then there is NO REASON to use T/t > > transfers. If your SIP devices do not support their own transfer > > option then either you didn't do enough research before you installed > > Asterisk or you were just too cheap when buying phones. > > > > Do you really need to park outbound external calls? > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura g729 call quality to PSTN
Thanks for the suggestion. Changing the RTP Packet Size in the Sipura to 40ms did improve the call quality "slightly", but still well below par compared to the Cisco 7960. In my ethereal captures, I did notice something interesting. While the RTP stream from the Cisco to asterisk seemed to have a 160 diffference in timestamps, the Sipura showed a 320 difference: Cisco: RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896 RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056 Sipura: RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771 RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091 On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns <[EMAIL PROTECTED]> wrote: > What is your sample size? > > I believe the 7960 supports 40ms (2 samples) per packet by default. > > Do you have an ethereal trace? Look at the timestamps between RTP packets if > you can't see/modify this setting. > > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Pedro > > Sent: Tuesday, February 15, 2005 6:30 PM > > To: Jeffrey Chan > > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN > > > > Actually the SPA-2100 supports 2 g729 channels which is why I bought > > it. Unfortunately, the call quality is just as poor on the 2100 as it > > is on the 2000. > > > > - Pedro > > > > > > On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan <[EMAIL PROTECTED]> > > wrote: > > > Is it just a bad implementation of g729 compression with the Sipura > > > > > > product line? > > > > > > > > > > > > > > That would be my guess too . why SPA-2000 supports G729 for one > > > channel only? no enough CPU power to code/decode G.729 for two > > > channels? > > > > > > Jeffey > > > > > > www.mutualphone.com > > > > > > > > > On Tue, 15 Feb 2005 16:31:59 -0500, Pedro <[EMAIL PROTECTED]> > wrote: > > > > uggg. > > > > > > > > Is anyone out there having any luck with the SPA-2000 or SPA-2100 > > > > using the g729 codec with decent call quality? > > > > > > > > > > > > On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler <[EMAIL PROTECTED]> > wrote: > > > > > > > > > > On Feb 14, 2005, at 1:25 PM, Pedro wrote: > > > > > > > > > > > > > > > > > Is it just a bad implementation of g729 compression with the > Sipura > > > > > > product line? > > > > > > > > > > > > > > > > That would be my guess. > > > > > > > > > > -mark > > > > > > > > > > -- > > > > > Mark Eissler, [EMAIL PROTECTED] > > > > > Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com > > > > > > > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@lists.digium.com > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura g729 call quality to PSTN
Forgot to mention that when I set the RTP Packet Size to 20ms that the difference was 160 (like the Cisco) but call quality was much worse. On Wed, 16 Feb 2005 15:36:44 -0500, Pedro <[EMAIL PROTECTED]> wrote: > Thanks for the suggestion. Changing the RTP Packet Size in the Sipura > to 40ms did improve the call quality "slightly", but still well below > par compared to the Cisco 7960. > > In my ethereal captures, I did notice something interesting. While > the RTP stream from the Cisco to asterisk seemed to have a 160 > diffference in timestamps, the Sipura showed a 320 difference: > > Cisco: > RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896 > RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056 > > Sipura: > RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771 > RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091 > > > On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns > <[EMAIL PROTECTED]> wrote: > > What is your sample size? > > > > I believe the 7960 supports 40ms (2 samples) per packet by default. > > > > Do you have an ethereal trace? Look at the timestamps between RTP packets if > > you can't see/modify this setting. > > > > > > > -Original Message- > > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > > [EMAIL PROTECTED] On Behalf Of Pedro > > > Sent: Tuesday, February 15, 2005 6:30 PM > > > To: Jeffrey Chan > > > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN > > > > > > Actually the SPA-2100 supports 2 g729 channels which is why I bought > > > it. Unfortunately, the call quality is just as poor on the 2100 as it > > > is on the 2000. > > > > > > - Pedro > > > > > > > > > On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan <[EMAIL PROTECTED]> > > > wrote: > > > > Is it just a bad implementation of g729 compression with the Sipura > > > > > > > product line? > > > > > > > > > > > > > > > > > That would be my guess too . why SPA-2000 supports G729 for one > > > > channel only? no enough CPU power to code/decode G.729 for two > > > > channels? > > > > > > > > Jeffey > > > > > > > > www.mutualphone.com > > > > > > > > > > > > On Tue, 15 Feb 2005 16:31:59 -0500, Pedro <[EMAIL PROTECTED]> > > wrote: > > > > > uggg. > > > > > > > > > > Is anyone out there having any luck with the SPA-2000 or SPA-2100 > > > > > using the g729 codec with decent call quality? > > > > > > > > > > > > > > > On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler <[EMAIL PROTECTED]> > > wrote: > > > > > > > > > > > > On Feb 14, 2005, at 1:25 PM, Pedro wrote: > > > > > > > > > > > > > > > > > > > > Is it just a bad implementation of g729 compression with the > > Sipura > > > > > > > product line? > > > > > > > > > > > > > > > > > > > That would be my guess. > > > > > > > > > > > > -mark > > > > > > > > > > > > -- > > > > > > Mark Eissler, [EMAIL PROTECTED] > > > > > > Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com > > > > > > > > > > > > > > > > > ___ > > > > > Asterisk-Users mailing list > > > > > Asterisk-Users@lists.digium.com > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > To UNSUBSCRIBE or update options visit: > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura g729 call quality to PSTN
FYI - Seems the latest firmware in conjunction with changing the packet size to 10ms improved the call quality to usable. The Cisco 7960 is stell superior, but now at least the SPA-2100 is acceptable (and with 2 working g729 channels including 3-way calling). On Wed, 16 Feb 2005 15:44:58 -0500, Pedro <[EMAIL PROTECTED]> wrote: > Forgot to mention that when I set the RTP Packet Size to 20ms that the > difference was 160 (like the Cisco) but call quality was much worse. > > > On Wed, 16 Feb 2005 15:36:44 -0500, Pedro <[EMAIL PROTECTED]> wrote: > > Thanks for the suggestion. Changing the RTP Packet Size in the Sipura > > to 40ms did improve the call quality "slightly", but still well below > > par compared to the Cisco 7960. > > > > In my ethereal captures, I did notice something interesting. While > > the RTP stream from the Cisco to asterisk seemed to have a 160 > > diffference in timestamps, the Sipura showed a 320 difference: > > > > Cisco: > > RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896 > > RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056 > > > > Sipura: > > RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771 > > RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091 > > > > > > On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns > > <[EMAIL PROTECTED]> wrote: > > > What is your sample size? > > > > > > I believe the 7960 supports 40ms (2 samples) per packet by default. > > > > > > Do you have an ethereal trace? Look at the timestamps between RTP packets > > > if > > > you can't see/modify this setting. > > > > > > > > > > -Original Message- > > > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > > > [EMAIL PROTECTED] On Behalf Of Pedro > > > > Sent: Tuesday, February 15, 2005 6:30 PM > > > > To: Jeffrey Chan > > > > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN > > > > > > > > Actually the SPA-2100 supports 2 g729 channels which is why I bought > > > > it. Unfortunately, the call quality is just as poor on the 2100 as it > > > > is on the 2000. > > > > > > > > - Pedro > > > > > > > > > > > > On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan <[EMAIL PROTECTED]> > > > > wrote: > > > > > Is it just a bad implementation of g729 compression with the Sipura > > > > > > > > product line? > > > > > > > > > > > > > > > > > > > > That would be my guess too . why SPA-2000 supports G729 for one > > > > > channel only? no enough CPU power to code/decode G.729 for two > > > > > channels? > > > > > > > > > > Jeffey > > > > > > > > > > www.mutualphone.com > > > > > > > > > > > > > > > On Tue, 15 Feb 2005 16:31:59 -0500, Pedro <[EMAIL PROTECTED]> > > > wrote: > > > > > > uggg. > > > > > > > > > > > > Is anyone out there having any luck with the SPA-2000 or SPA-2100 > > > > > > using the g729 codec with decent call quality? > > > > > > > > > > > > > > > > > > On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler <[EMAIL PROTECTED]> > > > wrote: > > > > > > > > > > > > > > On Feb 14, 2005, at 1:25 PM, Pedro wrote: > > > > > > > > > > > > > > > > > > > > > > > Is it just a bad implementation of g729 compression with the > > > Sipura > > > > > > > > product line? > > > > > > > > > > > > > > > > > > > > > > That would be my guess. > > > > > > > > > > > > > > -mark > > > > > > > > > > > > > > -- > > > > > > > Mark Eissler, [EMAIL PROTECTED] > > > > > > > Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com > > > > > > > > > > > > > > > > > > > > ___ > > > > > > Asterisk-Users mailing list > > > > > > Asterisk-Users@lists.digium.com > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > To UNSUBSCRIBE or update options visit: > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@lists.digium.com > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > To UNSUBSCRIBE or update options visit: > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura g729 call quality to PSTN
Actually - jitter does not seem to be the issue (sound is not garbled and does not drop out, it was just very low and "fuzzy"/"staticy" when not set to 10 ms). It is weird that I have to drop to 10ms, but I have tested some more and the general consenses from the people I have called said it sounds fine now with 10ms setting. Thanks for your help though. Here is the result set from the ethereal trace using 10ms (RTP stream sent from Sipura to asterisk): RTP Payload type=ITU-T G.729, SSRC=1783584165, Seq=7784, Time=156604121 RTP Payload type=ITU-T G.729, SSRC=1783584165, Seq=7784, Time=156604201 As you can see there is now a difference of 80 between the Time stamps (now to sound dumb, but it would be 80 what?) On Wed, 16 Feb 2005 19:42:19 -0700, Keith Burns <[EMAIL PROTECTED]> wrote: > Hmmm, that worked? > > Interesting that you can change the sample size to 10ms since the "standard" > is 20ms that most people don't go below. I know you *can* do below 20 but if > you are doubt the technical ability of the box it seems strange they are > capable of that. > > This seems to smack of bad de-jitter buffers on the egress gateway... are > you receiving 20ms sampled RTP ? > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Pedro > > Sent: Wednesday, February 16, 2005 3:20 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN > > > > FYI - Seems the latest firmware in conjunction with changing the > > packet size to 10ms improved the call quality to usable. The Cisco > > 7960 is stell superior, but now at least the SPA-2100 is acceptable > > (and with 2 working g729 channels including 3-way calling). > > > > > > On Wed, 16 Feb 2005 15:44:58 -0500, Pedro <[EMAIL PROTECTED]> > wrote: > > > Forgot to mention that when I set the RTP Packet Size to 20ms that the > > > difference was 160 (like the Cisco) but call quality was much worse. > > > > > > > > > On Wed, 16 Feb 2005 15:36:44 -0500, Pedro <[EMAIL PROTECTED]> > wrote: > > > > Thanks for the suggestion. Changing the RTP Packet Size in the Sipura > > > > to 40ms did improve the call quality "slightly", but still well below > > > > par compared to the Cisco 7960. > > > > > > > > In my ethereal captures, I did notice something interesting. While > > > > the RTP stream from the Cisco to asterisk seemed to have a 160 > > > > diffference in timestamps, the Sipura showed a 320 difference: > > > > > > > > Cisco: > > > > RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, > > Time=40666896 > > > > RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, > > Time=40667056 > > > > > > > > Sipura: > > > > RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, > > Time=434932771 > > > > RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, > > Time=434933091 > > > > > > > > > > > > On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns > > > > <[EMAIL PROTECTED]> wrote: > > > > > What is your sample size? > > > > > > > > > > I believe the 7960 supports 40ms (2 samples) per packet by default. > > > > > > > > > > Do you have an ethereal trace? Look at the timestamps between RTP > packets if > > > > > you can't see/modify this setting. > > > > > > > > > > > > > > > > -Original Message- > > > > > > From: [EMAIL PROTECTED] > [mailto:asterisk-users- > > > > > > [EMAIL PROTECTED] On Behalf Of Pedro > > > > > > Sent: Tuesday, February 15, 2005 6:30 PM > > > > > > To: Jeffrey Chan > > > > > > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > > > > > > Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN > > > > > > > > > > > > Actually the SPA-2100 supports 2 g729 channels which is why I > bought > > > > > > it. Unfortunately, the call quality is just as poor on the 2100 > as it > > > > > > is on the 2000. > > > > > > > > > > > > - Pedro > > > > > > > > > > > > > > > > > > On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan > > <[EMAIL PROTECTED]> > > >
Re: [Asterisk-Users] Sipura g729 call quality to PSTN
> That does not sound right at all. The difference between the two Time= > values should have been 10 (milliseconds). > > Did you reboot the Sipura after making the change? There are some values > in the Sipura that don't take effect until after the next reboot; I don't > have a clue whether this happens to be one of them. Yes - sipura was rebooted. Actually, the changes did seem to take affect even before the reboot (verified by call quality improvement and ethereal traces). So in your opinion, instead of 80, it should be a difference of 10? If so - then you are saying that the timestamp is in miliseconds? I am as puzzled as you - really does not seem logical, but call quality is finally decent and it does not seem to bother asterisk at all. Do you see any potential problems with this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage, broadvoice et al
Vonage, to my knowledge, does not let you connect your own SIP device to their service. They provide their own IAD. As for Broadvoice, I know people that have successfully deployed asterisk with many people sharing the same account. - Pedro On Fri, 18 Feb 2005 17:54:38 +0800, el Flynn <[EMAIL PROTECTED]> wrote: > Hi all, > > I'm just wondering about these VoIP services -- do you have to sign up one > account -per- client that will be using the service? I've got multiple > extensions behind my Asterisk box, and I want to be able to allow all my staff > to place calls via the provider. > > So if I sign up for one account, will multiple users behind my Asterisk box be > able to make calls, using that same account, at the same time? Or do these > providers typically only allow one call to be in place at any point in time? > > Thanks in advance. > > Flynn > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura g729 call quality to PSTN
Rich - thanks! Glad I am not the only one seeing this :) Would be very interested in your results. No problems that I see yet with these settings. On Fri, 18 Feb 2005 10:21:08 -0600, Rich Adamson <[EMAIL PROTECTED]> wrote: > > > That does not sound right at all. The difference between the two Time= > > > values should have been 10 (milliseconds). > > > > > > Did you reboot the Sipura after making the change? There are some values > > > in the Sipura that don't take effect until after the next reboot; I don't > > > have a clue whether this happens to be one of them. > > > > Yes - sipura was rebooted. Actually, the changes did seem to take > > affect even before the reboot (verified by call quality improvement > > and ethereal traces). > > > > So in your opinion, instead of 80, it should be a difference of 10? > > If so - then you are saying that the timestamp is in miliseconds? > > > > I am as puzzled as you - really does not seem logical, but call > > quality is finally decent and it does not seem to bother asterisk at > > all. Do you see any potential problems with this? > > I did a fair amount of experimenting this morning using a spa3000 with > g711 and g729 codecs. I'm more confused now then ever. I also used > ethereal to inspect timestamps, etc. > > spa3k(fxs) -> asterisk -> IAX(ITSP) -> pstn net -> analog phone > > The spa3k is running v2.0.13(GWg) firmware, and * is CVS-HEAD-02/13/05. > > The spa3k had a default RTP Packet Size of .030 (30 milliseconds) even > though the User Manual indicated that 20 milliseconds is the default. > Asterisk config is default at 20 milliseconds. > > I changed the spa3k rtp from .030 seconds, to .020 seconds for > consistency. Audio quality "seemed" to be better when using g711. > > Regardless of whether I used g711u or g729, the rtp timestamps were > always 160 difference between consequtive packets (as observed by > ethereal). > > Changing the spa3k rtp to .010 seconds yielded timestamps that were > always 80 difference between consequtive packets (same as you > observed). However, * -> spa3k continued to have 160 difference. > Audio quality seemed to improve another step, and the occasional > echo that we heard seemed to disappear. Pure guess is the smaller > rtp size is impacting the jitter buffer and/or echo canceller in > the spa3k. I'm going to run with these settings for a while to see > what the longer term impact/stability might be. > > Rich > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CODEC g723, g729, g711
Make sure you have the proper licenses to use the codecs: g729 http://www.digium.com/index.php?menu=asterisk_g729 g723 http://www.dspg.com/technology/LicensePricing.html On Fri, 18 Feb 2005 18:17:58 -0800, Nitesh Divecha <[EMAIL PROTECTED]> wrote: > Hello All, > > Any one has success with codec g723 & g729? > I am having extremely hard time to setup this codec. > The only codec worked is g711a/u. > > If I set g723 & g729 as first and second choice codec in my sip.conf, VM and > MeetMe stop working. > > Sip.conf > > [general] > port = 5060 ; Port to bind to (SIP is 5060) > bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) > disallow=all > ;allow=g273 > ;allow=g729 > allow=ulaw > allow=alaw > > #include sip_nat.conf > #include sip_additional.conf > > I am using Snom 220/200 and all are set to use g729. > > Thank you, > > Nitesh > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I have a odd question...
If you use the MySQL CDR add-on, you could just query the CDR DB for the numbers you are tracking. No need to add anything fancy. On Sat, 19 Feb 2005 21:42:31 +0100, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > > Hi all. > > I am going to do a simple "voting application" for a radiostation. > > The idea is to have listeners call in to vote on songs. > > What I want to do is to take a phonenumer for each song and present the > result on a simple webpage. > > Eg. > > To vote on song number one, call 555- > > To vote on song number two, call 555- etc etc. > > When the listener calls in, a playback tells him: "Thank you for voting > on song number one." > > And the numbers of calls on each number are presented on a webpage, or > in a textfile, easy for the showhost to see. > > How do I do this the simplest way ? > > I have a lot on phonenumbers that I can use, so that is not the problem. > > Shoud I execute some kind of script for each caller that increases the > numbers in a textfile ? Or how should I do ? > > My programmingskills aren't the best, so I would be greatful for any > help I can get. > > /Regards Mike. > > PS. Please answer offlist if possible.. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX channel unable to create
First off - change: exten => _2000,1,Dial(IAX/master:[EMAIL PROTECTED]/[EMAIL PROTECTED]) to: exten => _2000,1,Dial(IAX2/master:[EMAIL PROTECTED]/[EMAIL PROTECTED]) On Mon, 21 Feb 2005 13:00:39 -0500, kurt x <[EMAIL PROTECTED]> wrote: > I have two * boxes running two differnet versions of *. > Box A is running: > > Asterisk CVS-HEAD-07/14/04-16:28:29 built by > [EMAIL PROTECTED] on a i686 running Linux > > Box B is running: > > Asterisk 1.0.3 built by [EMAIL PROTECTED] on a i386 running FreeBSD > > I can make a IAX call from B to A but not from A to B. > When I try to make a call from A to B I get these messages: > > Feb 21 12:48:12 WARNING[-1233155152]: channel.c:1860 ast_request: No > channel type registered for 'IAX' > Feb 21 12:48:12 NOTICE[-1233155152]: app_dial.c:696 dial_exec: Unable > to create channel of type 'IAX' > Feb 21 12:48:14 WARNING[-1116300368]: chan_sip.c:673 retrans_pkt: > Maximum retries exceeded on call > [EMAIL PROTECTED] > for seqno 1 (Non-critical Response) > > My box A iax.conf: > [general] > port=5036 > bindport=5036 > bandwidth=low > allow=ulaw > disallow=lpc10 > jitterbuffer=no > tos=lowdelay > > [slave] > type=friend > secret=4435 > context=voice-mail > defaultip=192.168.2.232 > qualify=yes > > My Box A extension.conf > [voice-mail] > exten => _2000,1,Dial(IAX/master:[EMAIL PROTECTED]/[EMAIL PROTECTED]) > > My box B iax.conf > [general] > port=5036 > bindport=5036 > bandwidth=low > allow=ulaw > disallow=lpc10 > tos=lowdelay > > [master] > type=friend > secret=4435 > context=home > defaultip=192.168.1.2 > qualify=yes > > My Box B extension.conf > [home] > exten => _24xx,1,Dial(IAX2/slave:[EMAIL PROTECTED]/[EMAIL PROTECTED]) > > Thanks in advance > > Kurt > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP NEEDED ASTERISK AND MEDIATRIX 1102
Do yourself a favor and get a Sipura SPA-2100 - much easier to configure and the quality is better than the Mediatrix unit. First of all - do you have the Mediatrix Unit Manager software? If not, configuration will be nearly impossible. Secondly, you will need to configure the sip ports on the mediatrix to include "asterisk" as the realm. The other fields are pretty self explanatory (username, password, etc.). You will also want to turn off silence suppression as it is on by default. - Pedro On 25 Feb 2005 20:07:04 +0100, Edward Banfa <[EMAIL PROTECTED]> wrote: > Hello all, > > Hi I would like to know how to configure a Mediatrix 1102 box to work > with my asterisk box. I have analog phones that i would like to connect > to my Mediatrix box and then connect the Mediatrix box to my asterisk > box. My main problems come from the fact that I have limited experience > with usiing the two (asterisk and the mediatrix). I know how to use > sip.conf , but I am lost when it comes to mediatrix specific > configuration. I have search the archives but i have not gotten any > thing specific. > I would really appreciate any help that can be rendered to set me in the > right path. I am desperate here. > Thank you all in advance > > Edward > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zombie SIP channels
Ok - I finally found out what was causing the ZOMBIE channels. Now follow me on this one :) It appears that if you are using a Cisco 7960 and are on a call and want to transfer the call to another extension - if you press "more" and "Trnsfer" and dial the extension and you hit the Trnsfer button again before the extension answers, a ZOMBIE channel is created. If you use BlindXfer, it does not create the ZOMBIE channel. I have now informed my client that if they want to do a Blind Transfer, to use the BlindXfer softkey instead of the Trnsfer softkey or just use the # key to do a blind transfer. Now, I am running Asterisk CVS-v1-0-11/12/04-15:32:45. I would be interested in knowing if later versions of asterisk exhibited this same behavior. Any feedback would be appreciated. Thanks, Pedro On Fri, 11 Feb 2005 08:32:43 +0100, Florian Overkamp <[EMAIL PROTECTED]> wrote: > Hi, > > > -Original Message- > > Ok this is odd - caught it again twice today. The more I thought > > about what has changed on the server I realized that I was not using a > > timing device before, but am now using ztdummy. I if that could be > > causing the zombies? > > > > > http://bugs.digium.com/bug_view_page.php?bug_id=0002938 > > I don't think so, but who knows. The patch resolves a locking issue that may > or may not be timing-source dependant. I've seen the issue occur after call > transfers in scenario's where I used a few chan_local's. > > Do yourself a favour: > > - If you can, unload the ztdummy and test for a while. However, this may put > the issue to sleep - but it won't solve it! > - After that, load ztdummy again and apply the two lines in channel.c. Test > again. Good chance the issue will be gone. > > Report results here :) > > Florian > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP VoIP Provider problems
Sounds like you are having a codec issue with 2 of your providers. Make sure you find out what codecs are supported and that your config is set up accordingly. On Sun, 06 Mar 2005 00:14:05 +, w fm3 <[EMAIL PROTECTED]> wrote: > Hi > > Hope someone can help :) > > I am testing 4 PSTN termination providers. 3 SIP and 1 IAX > > IAX and 1 of the SIP providers work fine. > > Now the wierdness: > > 2 SIP providers I can only get oubound calls to ring at the destination and > then nothing more. 1 gets as far as SIP code 183 (and ringing on the src > handset ...yay) the other doesn't get past 100. > > Added to this inbound calls (PSTN->provider->asterisk->handset) work fine > 100% of the time. > > I have tried alot of config options from the wiki and lists but can't seem > to get any further. AFAIK from sip debug and the console it looks like > that the call is placed and then no further communication. Looks like they > might be using SER / CISCO GW at the VOIP Provider end. > Don't think it a open UDP port type thing. > > Cheers > > Walt > > PS Newbie > > _ > Express yourself instantly with MSN Messenger! Download today it's FREE! > http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP Firmware
Just a note that you will need to perform quite a few incremental upgrades to get to a current firmware version. So if you do get someone who will sell you the firmware, make sure you get the all of them. On Fri, 18 Mar 2005 22:04:29 -0500, Patrick M. Gray, Jr. <[EMAIL PROTECTED]> wrote: > > > > I got a new old stock Cisco 7960 from eBay and the warranty expired bay in > 2001 according to Cisco (I didn't even know they had VoIP in 2001 ;-) ). I > spoke with a wonderfully rude gentleman at Cisco who told me there was > nothing that could be done to get SIP firmware for the device, and would not > even entertain the possibility of purchasing said FW from Cisco. He > suggested I call a local reseller, and the single one I called was not > interested in helping me either with my "unsupported hardware." > > > > I'm using the 7960 to experiment with *, and was wondering if there are > alternative means to finding the firmware, or if the "out of the box" SCCP > firmware (I have version P003AM30) will work with *. I'm willing to pay any > "official resellers" a fair price for the F/W, but the attitude I received > from Cisco and the one reseller I contacted have me thinking this is a waste > of time. > > > > I'm using [EMAIL PROTECTED] and can seem to find any SCCP info, and don't want > to delve too deeply into this experiment if the phone is not going to work > reliably. > > > > Thanks for any help or pointers in the right direction. > > > > Pat > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 Released!
What jitterbuffer issues are you having with connecting to 1.0.x servers?On 11/17/05, Eric ManxPower Wieling <[EMAIL PROTECTED] > wrote:Asterisk guy wrote:> does it include the patch for VAD?> > ( dropping extra frame of G.729 since we already have a VAD frame at the end )It does not include several important things. It does not include a SIPjitter buffer. It does not include the ability to use Zaptel for timing of the RTP audio. It does not include VAD/CND support. As far as Iknow it also does not have the patch to make the new IAX2 jitterbufferwork correctly when connecting to a 1.0.x server.___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do you disable realtime?
Am I correct in assuming that if I am not running Realtime on my asterisk 1.2 server, the proper way to disable it is to remove the following 2 files: /usr/lib/asterisk/modules/pbx_realtime.so /usr/lib/asterisk/modules/app_realtime.so I am just testing out the default installation and am getting these errors on the console: Nov 21 12:05:29 ERROR[30656] res_config_mysql.c: MySQL RealTime: Failed to connect database server on . Check debug for more info. Nov 21 12:05:29 WARNING[30656] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. Any help will be appreciated. - Pedro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do you disable realtime?
Yeah - tried that. Here are 2 lines I have in my modules.conf file: noload => pbx_realtime.so noload => app_realtime.so For some reason, I still get the following in my logs even after a restart of Asterisk. Nov 21 13:17:08 ERROR[31192] res_config_mysql.c: MySQL RealTime: Failed to connect database server on . Check debug for more info. Nov 21 13:17:08 WARNING[31192] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. Any thoughts? - Pedro On 11/21/05, Alexander Lopez <[EMAIL PROTECTED]> wrote: It is a better practice to use a noload option in modules.conf. That way if and when you upgrade you wont need to remove them again they will just continue to not load Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of PedroSent: Monday, November 21, 2005 12:11 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] How do you disable realtime? Am I correct in assuming that if I am not running Realtime on my asterisk 1.2 server, the proper way to disable it is to remove the following 2 files:/usr/lib/asterisk/modules/pbx_realtime.so/usr/lib/asterisk/modules/app_realtime.soI am just testing out the default installation and am getting these errors on the console:Nov 21 12:05:29 ERROR[30656] res_config_mysql.c: MySQL RealTime: Failed to connect database server on . Check debug for more info.Nov 21 12:05:29 WARNING[30656] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug.Any help will be appreciated.- Pedro ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do you disable realtime?
Thanks Bruce - but the whole point I am trying to accomplish is that I don't want to use Realtime and don't want asterisk to try to establish the connection. Was just chatting in IRC about this and it seems that Realtime may not be able to be truly disabled (not sure how accurate that is, but that was what I was told). Basically I just want to have asterisk load without those 2 errors popping up on the console and in the logs.On 11/21/05, Bruce Ferrell <[EMAIL PROTECTED]> wrote: Check the mysql logs. I would suspect from this one of several things:1.) the userid/password is incorrect.on the db host use the command lin e mysql client like so: mysql -h localhost -u -p you'll be prompted for a password. If that works, go to the next possible problem2.) the userid doesn't have correct permissions to the DB from the mysql client, issues the use command to try to access the realtime DB. if that works, go to the next possible problem.3.) the userid is not permitted from the host the asterisk box is on as the mysql superuser look at mysql.user to see what hosts are permitted access by the asterisk userid/password. If you have to add a host, be sure to issue the flush priviledges commandPedro wrote:> Yeah - tried that. Here are 2 lines I have in my modules.conf file:>> noload => pbx_realtime.so> noload => app_realtime.so>> For some reason, I still get the following in my logs even after a> restart of Asterisk.> > Nov 21 13:17:08 ERROR[31192] res_config_mysql.c: MySQL RealTime: Failed> to connect database server on . Check debug for more info.> Nov 21 13:17:08 WARNING[31192] res_config_mysql.c: MySQL RealTime: > Couldn't establish connection. Check debug.>> Any thoughts?>> - Pedro>> On 11/21/05, Alexander Lopez <[EMAIL PROTECTED]> [EMAIL PROTECTED]>> wrote:>> It is a better practice to use a noload option in modules.conf. That> way if and when you upgrade you wont need to remove them again they > will just continue to not load>> Alex>>> > From: [EMAIL PROTECTED]> [EMAIL PROTECTED]>> [mailto: [EMAIL PROTECTED]> [EMAIL PROTECTED]>] On Behalf Of Pedro> Sent: Monday, November 21, 2005 12:11 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion> Subject: [Asterisk-Users] How do you disable realtime?>> Am I correct in assuming that if I am not running Realtime on my > asterisk 1.2 server, the proper way to disable it is to remove> the following 2 files:>> /usr/lib/asterisk/modules/pbx_realtime.so> /usr/lib/asterisk/modules/app_realtime.so >> I am just testing out the default installation and am getting> these errors on the console:>> Nov 21 12:05:29 ERROR[30656] res_config_mysql.c: MySQL RealTime: > Failed to connect database server on . Check debug for more info.> Nov 21 12:05:29 WARNING[30656] res_config_mysql.c: MySQL> RealTime: Couldn't establish connection. Check debug.>> Any help will be appreciated.>> - Pedro >>> ___> --Bandwidth and Colocation sponsored by Easynews.com> < http://Easynews.com> -->> Asterisk-Users mailing list> Asterisk-Users@lists.digium.com Asterisk-Users@lists.digium.com>> http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users>>>> >> ___> --Bandwidth and Colocation sponsored by Easynews.com -->> Asterisk-Users mailing list> Asterisk-Users@lists.digium.com> http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do you disable realtime?
Olle, Yep - was actually replying to this as I got your message - I was searching for modules that had realtime in the name (did not see the res_config_mysql.so file). Setting the noload => res_config_mysql.so in modules.conf took care of the issue I was having. Thanks for your prompt response! -PedroOn 11/21/05, Olle E Johansson <[EMAIL PROTECTED]> wrote: Pedro wrote:> Yeah - tried that. Here are 2 lines I have in my modules.conf file:>> noload => pbx_realtime.so> noload => app_realtime.so>> For some reason, I still get the following in my logs even after a > restart of Asterisk.>> Nov 21 13:17:08 ERROR[31192] res_config_mysql.c: MySQL RealTime:> Failed to connect database server on . Check debug for more info.> Nov 21 13:17:08 WARNING[31192] res_config_mysql.c: MySQL RealTime: > Couldn't establish connection. Check debug.>> Any thoughts?>> - Pedro>> On 11/21/05, *Alexander Lopez* <[EMAIL PROTECTED] > [EMAIL PROTECTED]>> wrote:>> It is a better practice to use a noload option in modules.conf.> That way if and when you upgrade you wont need to remove them > again they will just continue to not load>> Alex>>> > *From:* [EMAIL PROTECTED]> [EMAIL PROTECTED]>> [mailto: [EMAIL PROTECTED]> [EMAIL PROTECTED]>] *On Behalf> Of *Pedro> *Sent:* Monday, November 21, 2005 12:11 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion> *Subject:* [Asterisk-Users] How do you disable realtime?>> Am I correct in assuming that if I am not running Realtime on > my asterisk 1.2 server, the proper way to disable it is to> remove the following 2 files:>> /usr/lib/asterisk/modules/pbx_realtime.so> /usr/lib/asterisk/modules/app_realtime.so >> I am just testing out the default installation and am getting> these errors on the console:>> Nov 21 12:05:29 ERROR[30656] res_config_mysql.c: MySQL> RealTime: Failed to connect database server on . Check debug > for more info.> Nov 21 12:05:29 WARNING[30656] res_config_mysql.c: MySQL> RealTime: Couldn't establish connection. Check debug.>> Any help will be appreciated. >Realtime is implemented in several places. PBX_realtime is the realtimeswitch, app_realtime is an application.res_config_mysql.c/so is the realtime driver for Mysql databases. So no,you are not correct. You have not removed all the modules that involve realtime.On the other hand, the easiest way to disable realtime is not to enableit in the configuration file, extconfig.confYes, it's a strange name, but there are historical reasons for it :-) /O___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec that quality does not get affect *much* against packet loss
I think you are thinking of iLBC: http://www.voip-info.org/wiki-iLBC Be aware that this codec is known to be pretty CPU intensive to accomplish its compression. - PedroOn 11/22/05, Sam Tam <[EMAIL PROTECTED]> wrote: I think I have heard in the past that someone mentioned to me there is acodec that does not getting affected much because of packet loss.Is there such thing?Sam___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.10
I noticed that asterisk.org now has asterisk and zaptel downloads for version 1.0.10 but libpri, addons and sounds are still showing a 1.0.9 version number. Just wondering for those using the 1.0.x versions of asterisk instead of the 1.2 versions - will libpri, addons and sounds be updated to match the 1.0.10 version or will 1.0.9 be the final release of those packages? - Pedro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk 1.0.10
Note - looks like the answer to this was posted out of *date* sequence on asterisk.org (it is below the 1.2.0 release notice): direct from asterisk.org homepage: "Version 1.0.10 has been released of Asterisk and Zaptel. Libpri, Asterisk-addons, and Asterisk-sounds contain no changes, so they have not been updated. It is very likely that this will be the final release of the 1.0 branch of Asterisk. Users are strongly encouraged to begin upgrading to version 1.2. Thanks!" On 11/22/05, Pedro <[EMAIL PROTECTED]> wrote: I noticed that asterisk.org now has asterisk and zaptel downloads for version 1.0.10 but libpri, addons and sounds are still showing a 1.0.9 version number. Just wondering for those using the 1.0.x versions of asterisk instead of the 1.2 versions - will libpri, addons and sounds be updated to match the 1.0.10 version or will 1.0.9 be the final release of those packages? - Pedro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi Phones
The UTStarcom F1000 with the latest firmware (3.10st) has improved sound volume over the default firmware shipped with the units. Also, TFTP configuration works well so you don't have to configure the units with the keypad. You will need to get the configuration compiler from your vendor and be aware that the default encryption key should be set to NULL rather than F1000 as stated in the docs when compiling your config. At first I was not sure how I would like a WiFi phone because I figured it would sound bad, but I have been very impressed with the quality of the F1000. We have now added it to our VoIP product offerings. - Pedro http://www.traci.netOn 10/8/05, Cory Andrews <[EMAIL PROTECTED]> wrote: The F3000 is not anticipated to be available for distribution until lateDecember/January, FYI. Cory AndrewsSenior Partner+++VOIPSupply.com454 Sonwil DriveBuffalo, NY 14225+++voice - 716.630.1555 X22email - [EMAIL PROTECTED] fax - 716.630.1548Denis Galvão - iSolve wrote:> Wait for the next UTStarCom version... Called F3000, Im not sure, but> something like that.>> It will have better battery performance and will have 802.11g> support, and many other improvements. It will be available soon.>> Denis.>>>> On 07 de out de 2005, at 00:54, Andy Hamilton wrote:>>>> Anyone have good words to say about any of the WiFi handsets currently >>> available?>>>>>>> The UTStarCom F1000 (an 802.11b device) works pretty well. It's about>> half the $$$ of a Cisco 7920 (which are also pretty nice), but it>> seems like most of the config is done from the keypad. There is a TFTP >> option, but it seems that isn't quite perfect. You could check the>> manual (I programmed the unit without that, except to find that the>> default password is 88).>>>> The unit, I'm guessing, was designed somewhere in Asia, and the >> language translation shows it a little bit. Sound quality seems pretty>> good for the few calls I've passed through it. I only have one AP in>> my house, so I can't comment on roaming. The headset for my cell phone >> is stereo, and I think the phone would be most happy with a standard 3>> conductor plug, but I imagine a headset on a phone is a headset on a>> phone.>>>> The keypad is a touch small, and sometimes I hit the wrong key (and my >> fingers aren't terribly fat). I also seemed to have a problem>> transferring calls (using the built in transfer function -- # should>> still work). Despite many vendors' pages saying that it does 802.1x>> authentication, it sure looks like WEP is the only available>> "security" option.>>>> Overall: I would recommend purchasing one, for testing at the very>> least. >> They are well priced and of good quality.>>>> Battery life seems to be pretty good, too.>>>> -A>> ___>> --Bandwidth and Colocation sponsored by Easynews.com -->>>> Asterisk-Users mailing list>> Asterisk-Users@lists.digium.com>> http://lists.digium.com/mailman/listinfo/asterisk-users>> To UNSUBSCRIBE or update options visit:>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>> ___> --Bandwidth and Colocation sponsored by Easynews.com -->> Asterisk-Users mailing list > Asterisk-Users@lists.digium.com> http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users>>___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] res_musiconhold.c: Music on Hold class 'default' already exists
I just installed asterisk 1.2 rc2 and ran a 'make samples' and asterisk starts just fine with no errors in the logs. However, if I issue a reload I get the following: Nov 15 17:08:22 WARNING[27009] res_musiconhold.c: Music on Hold class 'default' already exists It is almost like the previous musiconhold process was not stopped (guessing here as I am not a programmer)? Does this make sense? Has anyone else seen this? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound calling number problem
Hello, We have a E400P and when we make a call we can´t change de calling number in my ISDN primary number range. We have 5 number over them and even have the first one identity in the called party. The "pri debug" says: "> Calling Number (len= 4) [Ext: 0 TON: Unknown Number Type (0) ..." What is wrong ? Regards, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax can't pass trough alaw
Hi, We have a e405p with a external Euro-isdn PRI-ISDN net interface from Telco connected. We tried to send a fax to another machine with a TDM400P. We use IAX2 with G711-alaw codec. Both fax machines connect, but have error in transfer. We use asterisk CVS-02/01/04. Which can be the problem ?. What can I do to find the problem ? Thanks. Regards, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax problem
Hi, We have a machine with an *'s with Digium TDM400P and connected wit other machine with *'s an TDM400P too. Well, I have a fax connected to each machine, and the protocol in the middle is IAX2 alaw. The fax between two fax, on in each machine, not work. The fax answer, but error in comm. Which can be the problem ?. What can I do to find the problem ? Thanks, in advance, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy error
Hi,
When have a incoming call from E1 to a extension FXS, and this extension is
busy, the incoming call recive ring tone, and it is wrong. What can I do?
Thanks in advance
Pedro
Here is the trace:
asterisk-1*CLI>
< Protocol Discriminator: Q.931 (8) len=41
< Call Ref: len= 2 (reference 66/0x42) (Originator)
< Message type: SETUP (5)
< Sending Complete (len= 4)
< Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
< Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
< Ext: 1 User information layer 1: A-Law (35)
< Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred
Dchan: 0
<ChanSel: Reserved
< Ext: 1 Coding: 0 Number Specified Channel Type:
3
< Ext: 1 Channel: 15 ]
< Calling Number (len=13) [ Ext: 0 TON: Unknown Number Type (0) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
< Presentation: Presentation allowed of network
provided number (3) '666343536' ]
< Called Number (len=12) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '917151314' ]
-- Making new call for cr 66
-- Processing Q.931 Call Setup
-- Processing IE 33 (Sending Complete)
-- Processing IE 4 (Bearer Capability)
-- Processing IE 24 (Channel Identification)
-- Processing IE 108 (Calling Party Number)
-- Processing IE 112 (Called Party Number)
> Protocol Discriminator: Q.931 (8) len=10
> Call Ref: len= 2 (reference 32834/0x8042) (Terminator)
> Message type: CALL PROCEEDING (2)
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
>ChanSel: Reserved
> Ext: 1 Coding: 0 Number Specified Channel Type:
3
> Ext: 1 Channel: 15 ]
> Protocol Discriminator: Q.931 (8) len=14
> Call Ref: len= 2 (reference 32834/0x8042) (Terminator)
> Message type: ALERTING (1)
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
>ChanSel: Reserved
> Ext: 1 Coding: 0 Number Specified Channel Type:
3
> Ext: 1 Channel: 15 ]
> Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0:
0 Location: Private network serving the local user (1)
> Ext: 1 Progress Description: Inband
information or appropriate pattern now available. (8) ]
-- Accepting call from '666343536' to '917151314' on channel 15, span 2
-- Executing Macro("Zap/19-1", "stdexten|72000|Zap/1") in new stack
-- Executing Dial("Zap/19-1", "Zap/1|200") in new stack
Apr 23 21:22:02 NOTICE[1130522]: app_dial.c:554 dial_exec: Unable to create
channel of type 'Zap'
== Everyone is busy at this time
-- Executing Busy("Zap/19-1", "") in new stack
< Protocol Discriminator: Q.931 (8) len=13
< Call Ref: len= 2 (reference 66/0x42) (Originator)
< Message type: STATUS (125)
< Cause (len= 3) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Public network serving the local user (2)
< Ext: 1 Cause: Invalid information element contents
(100), class = Protocol Error (6) ]
< Cause data 0: 01 (1)
< Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state:
Call Received (7)
-- Processing IE 8 (Cause)
-- Processing IE 20 (Call State)
asterisk-1*CLI>
At this point the incoming call have ring indication , and nothig is
ringing, is *'s sending busy ?.
After hangup the incoming call (externel origin):
asterisk-1*CLI>
< Protocol Discriminator: Q.931 (8) len=9
< Call Ref: len= 2 (reference 66/0x42) (Originator)
< Message type: DISCONNECT (69)
< Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
User (0)
< Ext: 1 Cause: Normal Clearing (16), class = Normal Event
(1) ]
-- Processing IE 8 (Cause)
-- Channel 15, span 2 got hangup
== Spawn extension (macro-stdexten, s, 102) exited non-zero on 'Zap/19-1'
in macro 'stdexten'
== Spawn extension (default, 917151314, 1) exited non-zero on 'Zap/19-1'
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
peerstate Disconnect Request
> Protocol Discriminator: Q.931 (8) len=9
> Call Ref: len= 2 (reference 32834/0x8042) (Terminator)
> Message type: RELEASE (77)
> Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user (1)
> Ext: 1 Cause: Normal Clearing (16), class = Normal Event
(1) ]
-- Hungup 'Zap/19-1'
< Protocol Discriminator: Q.931 (8) len=9
< Call Ref:
RE: [Asterisk-Users] call initiation
Roger, Maybe you are using extensions like "_9." try to put de complete number in your estension.conf ej; exten => _9XXX,1,Dial(. exten => 101,1,Dial(Zap/1) in that case send congestion if the 3 digits extensions are not in extensions.conf. Regards, Pedro J. Vela Ruiz Director Técnico Bomonte Tecnologías SL (BoMonTec) Tel. 902 141 181 -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Roger Enviado el: viernes, 23 de abril de 2004 21:39 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] call initiation Users withing the office can dial a 3 digit extension and that will ring a phone. The problem I'm running into is you have to press xxx then press 'send or 'dial'. The pbx doesn't recognize a 3 digit number as an internal extension and automatically dial it the user has to initiate that call. Asterisk automatically initiates calls w/ 9+7 digits and LD calls, 9+1+areacode+number. How would you tell the PBX try an extension once and 3 digits have been pressed. The exception being 9 as that gives a outside line. -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor[EMAIL PROTECTED] Rockford, IL 61101 815-968-9888 x101 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax problem
I use ulaw and the problem is the same, any more suggestions? Thanks -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Sam Bingner Enviado el: sábado, 24 de abril de 2004 6:11 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Fax problem Use ulaw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Vela Sent: Friday, April 23, 2004 7:52 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Fax problem Hi, We have a machine with an *'s with Digium TDM400P and connected wit other machine with *'s an TDM400P too. Well, I have a fax connected to each machine, and the protocol in the middle is IAX2 alaw. The fax between two fax, on in each machine, not work. The fax answer, but error in comm. Which can be the problem ?. What can I do to find the problem ? Thanks, in advance, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] quadBRI telco part hungs
Hi people, We have configured and working quadBRI in NT and TE mode. In TE part have pri_cpe_ptmp signalling and quadBRI leds are green. When asterisk in in verbose >=2 said "D-Channel on span 3 down" and immediately "D-Channel on span 3 up" maybe something is wrong, because in two hours the asterisk an quadBRI down responds to make calls or pickup. The system is running and we need reboot the system to reset the quadBRI because dont respond with unload and load modules. We are using RC20. If we put the pri debug on nothings appear. We need put the system stable, what can I do?, what information you need to help us?, very thanks. Regards, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] quadBRI & ISDN telephone
Hello, We have a quadBRI in NT mode with bri_cpe_ptmp signalling and when connect a ISDN telephone to this nothings happen. What can I do? My config files are this: Zaptel.conf: loadzone=es defaultzone=es # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,1,3,ccs,ami span=3,1,3,ccs,ami span=4,1,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 Zapata.conf: [channels] ; Default language language=es ; switchtype = euroisdn pridialplan = local prilocaldialplan = local context=default group = 1 signalling = bri_net_ptmp channel => 1-2,4-5,7-8,10-11 Thanks, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quadBRI and UK ISDN2e
Hi, Maybe I have similar problem. I have a Junghanns.net quadBRI PCI Card in wiht Telefonica ISDN BRI line, and we have in zapata.conf usecallerid=yes and hidecallerid=no, but we have not the caller ID. Can I make some configuration to solve this? Thanks, Pedro -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Jon Fautley Enviado el: jueves, 08 de abril de 2004 9:51 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] quadBRI and UK ISDN2e Stephen Karrington wrote: > Which brand of card did you get? The Junghanns.net quadBRI PCI Card. Just been back through BT order processing and told them to put "Caller Display" (as they call it) on the line, which they said they've done... getting fairly certain it's not a BT issue now :| Any help greatly appreciated, Thanks, Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] quadBRI and CallerID
Hi, I have a Junghanns.net quadBRI PCI Card with Telefonica ISDN BRI line, and we have in zapata.conf usecallerid=yes and hidecallerid=no, but we have not the caller ID. Can I make some configuration to solve this? Thanks, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cdr-csv / Zaptel BRI / full number not stored (missing leading zeroes)
Hi Alf, Have you got a Junghanns.net quadBRI PCI Card ? If yes, Have you received CallerID number ? How you have got configured zaptel and zapata ? I´m collapssed at this point, thaks in advance, Pedro PD: maybe... around your question, are you using cdr in csv or in mysql?, if you are using mysql check if your calling party field are text type, because if are digits the left zeroes are stripped out. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Frederic Olivie Enviado el: domingo, 16 de mayo de 2004 16:22 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] cdr-csv / Zaptel BRI / full number not stored (missing leading zeroes) Hi, I'm using a ZaptelBRI card. It works fine. But I have a small problem with call logs. The leading zeroes of the external calling party are not stored (e.g. : 0140302010 will be stored as 140302010). Same for international numbers for which "00" will be stripped out. I would not mind if the cdr record would give me an indication of the call's origin (national or international), but it does not. The goal here is to implement a basic "missed call" web service that would allow my users to generate a call back. -- Frédéric Olivié (Alf) @ Club-Internet « Don't SCREAM, It hurts my eyes ! — Ne CRIEZ pas, ça fait mal aux yeux ! » —Alf, March 2001 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP connection
I need help to make a conection form FWD to my pbx, I can receive a call from PSTN for a FXo card but know I need to receive call via IP form FWD I have activate hte IAX on freeworlddialup but does not work I can't make or receive calls. I virtually new in this can please somebody help me. thanks, scorpionny ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hide caller id
Hi, We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using restrictcid=yes and doesn´t work. What can I do, thaks Pedro -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de [EMAIL PROTECTED] Enviado el: miércoles, 31 de marzo de 2004 12:00 Para: [EMAIL PROTECTED] Asunto: Asterisk-Users digest, Vol 1 #3275 - 3 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Re: AW: [Asterisk-Users] CAPI problems when loading chan_capi.so (Martin Mielke) 2. RE: RxFax/spandsp: not disconnecting (Reynaldo Simbulan) 3. Re: Asterisk and picoCell GSM Base Stations (Simon Anderson) --__--__-- Message: 1 Date: Wed, 31 Mar 2004 10:42:23 +0200 From: Martin Mielke <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: AW: [Asterisk-Users] CAPI problems when loading chan_capi.so Reply-To: [EMAIL PROTECTED] Hallo Sacha, :-P Sascha Knific wrote: >Hi > > > >>capiinfo gives: >>--- >>capi not installed - No such device or address (6) >>--- >> >> > >It´s not just about installing the apropriate package but you have to load >the capi kernel module for your isdn card. > >The module to load on boot time is set in /etc/isdn/capi.conf (on Debian). >You have to check how it´s done on your distro (I presume RedHat or SuSE). > > I use SuSE 9.0 Pro. I don't see any capi.conf - the only similar thing is /etc/capisuite/capisuite.conf but I don't know if we're talking about the same file... The module is loaded at system boot: --- pbx:~ # dmesg | grep -i capi capifs: Rev 1.1.4.1 CAPI-driver Rev 1.1.4.1: loaded capi20: started up with major 68 kcapi: capi20 attached capi20: Rev 1.1.4.2: started up with major 68 (middleware+capifs) --- I hope it's the right one... >You can load the module manually. For a AVM Fritz!Card PCI you would do: >"modprobe fcpci" > > The system has an Eicon Diva Server BRI 2M... and by now I can't find an specific module... Martin --__--__-- Message: 2 From: "Reynaldo Simbulan" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Date: Wed, 31 Mar 2004 18:39:30 +1000 Subject: [Asterisk-Users] RE: RxFax/spandsp: not disconnecting Reply-To: [EMAIL PROTECTED] Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag In state 8 Disconnecting Changed from phase 3 to 7 *CLI> show channels Channel (ContextExtensionPri ) State Appl. Data Zap/1-1 (faxservers3 ) Up RxFAX /var/lib/asterisk/fax/new/20040329-234801-0755965128.tif 1 active channel(s) - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, March 31, 2004 6:27 PM Subject: Asterisk-Users digest, Vol 1 #3273 - 10 msgs > Send Asterisk-Users mailing list submissions to > [EMAIL PROTECTED] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > >1. Re: Asterisk Security Audit? (Steven Critchfield) >2. DTMF Detection Problem (Ron McMillin) >3. Re: Caller entered digits ignored during wait (Tilghman Lesher) >4. Re: Sipcall.co.uk & [*] (Dave Cotton) >5. Re: IAX2 trunk mode over satellite ([EMAIL PROTECTED]) >6. Register vith SIP provider from behind NAT (Simon Brown) >7. Can't talk on Cisco VIP 30 using Chan Skinny (Dean) >8. Re: Caller entered digits ignored during >wait (Stig Andersson) >9. RE: Exception flag set - snom200 (jc) > > -- __--__-- > > Message: 1 > Subject: Re: [Asterisk-Users] Asterisk Security Audit? > From: Steven Critchfield <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Date: Tu
RE: [Asterisk-Users] hide caller id
Yes, my phone company has enabled the Caller ID hiden possibility, thats because with a Panasonic PBX works fine but with Asterisk not. Thanks for your aproach, what can I do now? Regards, Pedro -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Manuel Wenger Enviado el: viernes, 11 de junio de 2004 10:36 Para: [EMAIL PROTECTED] Asunto: R: [Asterisk-Users] hide caller id Before starting to look at the problem in Asterisk, make sure that your phone company has enabled the "selective CLIR" feature. Otherwise the phone exchange will simply ignore your request to hide CLIP. Regards Manuel -Messaggio originale- Da: Pedro Vela [mailto:[EMAIL PROTECTED] Inviato: venerdì, 11. giugno 2004 08:56 A: [EMAIL PROTECTED] Oggetto: [Asterisk-Users] hide caller id Hi, We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using restrictcid=yes and doesn´t work. What can I do, thaks Pedro ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] quadBRI FAX problem
Hello, We have a Asterisk CVS-HEAD-08/13/14-12:00:00-BRI-stuffed-0.1.0-RC4a and we have problem with fax. zapata.conf: group = 1 signalling = bri_net channel => 1,2 channel => 4-5 group = 2 signalling = bri_cpe channel => 7-8 channel => 10-11 Before install asterisk we have a Panasonic PBX directly to ISDN lines and voice and fax work fine. Now, we have between ISDN lines and Panasonic PBX the Asterisk, and voice is ok but fax doesn´t work fine. What can I do? Thanks, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quadBRI FAX problem
Hi, Thanks Tim, we try this and works fine at first page but when the page graphic dense or more than one page, we have an error. Un saludo, Pedro -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Robinson Tim-W10277 Enviado el: miércoles, 13 de octubre de 2004 14:02 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] quadBRI FAX problem Pedro You probably need to disable echo cancel when bridged. Can't recall the exact zapata.conf line. I had problems faxing through Asterisk until I disabled echo cancelling on bridged Zaptel calls. Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Vela Sent: 13 October 2004 10:12 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] quadBRI FAX problem Hello, We have a Asterisk CVS-HEAD-08/13/14-12:00:00-BRI-stuffed-0.1.0-RC4a and we have problem with fax. zapata.conf: group = 1 signalling = bri_net channel => 1,2 channel => 4-5 group = 2 signalling = bri_cpe channel => 7-8 channel => 10-11 Before install asterisk we have a Panasonic PBX directly to ISDN lines and voice and fax work fine. Now, we have between ISDN lines and Panasonic PBX the Asterisk, and voice is ok but fax doesn´t work fine. What can I do? Thanks, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400 synch issue
Hello, I have teh same problem with: QuadBRI -> * -> TDM400 -> Modem Thanks in advance for your help. Regards, Pedro -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Carl Sempla Enviado el: jueves, 23 de septiembre de 2004 3:56 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] TDM400 synch issue Hello, I have the following configuration : E100P -> * -> TDM400 -> Modem When I receive FAXes, about 20% of them are corrupted : pages are not always complete. If the fax is complex or with numerous pages, it's usually a mess. Before that, I was using spandsp with success. Unfortunately it's too picky with some broken fax (training failed). Since this failure only occurs with the same set of faxes and it's reproducible, I'm confident about my E100P configuration. That's why I suspect frame slips on the TDM400 side. How can I solve this issue ? It's a dual p3, highly idle without IRQ sharing : 20: 602869735 602487558 IO-APIC-level wctdm 21: 602485799 602863515 IO-APIC-level t1xxp zaptel.conf : span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 fxoks=32-35 zapata.conf : [channels] faxdetect=incoming context=default switchtype=euroisdn signalling=fxo_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no signalling=pri_cpe context=incoming channel => 1-15,17-31 signalling=fxo_ks context=internal channel => 32-33 Thanks, -- Carl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Type of T1 for T100P card
I'm currently setting up a PBX system using the T100P card, and was wondering if it can handle the 2-way trunk type of T1s. Do 2-way trunk T1s use RBS signaling? Please excuse my ignorance, I have mostly dealt with PRI B and D channel type of T1s. Thanks Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pros and cons on SIP vs H.323 vs MGCP
Just trying to get a feel for how these protocols have progressed and what is recommended from experience. Also, what phones work the best. Thanks for the info Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX between two *
Hi, I’m trying to connect two * with IAX. I can’t call from one * SIP Extension to another * SIP extensions. Somebody have a sample about how I can config IAX. Thanks, Pedro. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ERROR: cannot load module kernelcapi
Trying to get asterisk work but...
What might be the problem. I get the following error when I run capiinit.
System is fedora core2 with kernel 2.6.5-1.358.
ERROR: cannot load module kernelcapi
lsmod shows that kernelcapi should be ok.
[EMAIL PROTECTED] root]# lsmod
Module Size Used by
snd_pcm_oss40740 0
snd_pcm68872 1 snd_pcm_oss
snd_page_alloc 7940 1 snd_pcm
snd_timer 17156 1 snd_pcm
fcpci 499096 0
capi 12992 0
capifs 3720 2 capi
kernelcapi 38688 2 fcpci,capi
snd_mixer_oss 13824 1 snd_pcm_oss
snd38372 4 snd_pcm_oss,snd_pcm,snd_timer,snd_mixer_oss
soundcore 6112 1 snd
parport_pc 19392 1
lp 8236 0
parport29640 2 parport_pc,lp
autofs410624 0
rfcomm 27164 0
l2cap 16004 5 rfcomm
bluetooth 33636 4 rfcomm,l2cap
sunrpc101064 1
3c59x 30376 0
ipt_REJECT 4736 1
ipt_state 1536 1
ip_conntrack 24968 1 ipt_state
iptable_filter 2048 1
ip_tables 13440 3 ipt_REJECT,ipt_state,iptable_filter
floppy 47440 0
sg 27552 0
scsi_mod 91344 1 sg
microcode 4768 0
dm_mod 33184 0
uhci_hcd 23708 0
mga89008 2
ipv6 184288 8
ext3 102376 1
jbd40216 1 ext3
capiinfo shows the following:
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.101-02 (49.18)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
B1 protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 for fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
B3 protocols support: 0x80bf
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
T.30 for fax group 3 with extensions
Modem
0100
0200
3900
1f010040
1b0b
bf80
0101 0002
Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
Any help would be highly apreciated.
~pete
___
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Hakupalvelut aina mukanasi - kännykässä: http://www.eniro.fi/mobiili/wap/
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[Asterisk-Users] CISCO IP Conference Station
Hi, Somebody have any idea how I can config a CISCO IP CONFERENCE STATION Model 7935 that work with * . Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SKINNY with Cisco IP Conference 7935
Hi, I have one Cisco IP Conference 7935. I’m trying to config the SKINNY Protocol. I config the skinny.conf file same like sample for Cisco 7910. When somebody call me my phone ring and answer the call but I can’t hear anything, But the other people is hearing me very good. When I try to call somebody the * show me an error : RECEIVED UNKNOWN MESSAGE TYPE: 4 Somebody have a “skinny.conf” sample file for Cisco 7935 or any trick to fix this problem Thanks, Pedro. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] park announcement not working Help!
So I basically have park working but when the call gets parked it doesn't announce the line it parked on. How can I get this to work? Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco IP Conference 7935
Hi, I have one Cisco IP Conference 7935. Somebudy have any idea how I can config this phone to work woth “*”. My “*” server is now working with GrandStream Phone and X-Lite SoftPhone, I need to add this Cisco 7935 but I don’t know how I can convert to SIP. Thanks, Pedro Mansilla Consulting Integrating Services [EMAIL PROTECTED] T: (305) 567-0090 Ext. 24 F: (305) 567-0200 http://www.cis-international.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outgoing call queue.
Hi all, is it possible to make a queue for outgoing calls? That's for preventing "Device '/dev/ttyI 0' is busy" error when having only one line to dialout and many files in /var/spool/asterisk/outgoing folder. So it would call only one call at the time and when it's done it would move to next. Thanx in advance. ~pete ___ Etsi ystävien ja tuttujen yhteystiedot: http://henkilot.eniro.fi/ Hakupalvelut aina mukanasi - kännykässä: http://www.eniro.fi/mobiili/wap/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] alcatel omnipcx
Hi, can anyone tell me how i do a sip trunk between an asterisk and a alcatel omnipcx pbx with sip support tx, Pedro Santos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sound files
Hello, Can i identify the sound files that are played in the asterisk console? I defined the verbose to 100 but i can not see the sound files that are played in some situations... :( For example, I need to know what files are played for the message: "Extension xxx is unavailable...". The goal is to translate that to Portuguese (pt_pt)... Thanks in advance, PS. ___ --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] muscionhold error message
hi there guys! how can I eliminate this message? [May 11 11:00:46] WARNING[7039]: res_musiconhold.c:506 monmp3thread: Unable to spawn mp3player [May 11 11:09:06] WARNING[7039]: res_musiconhold.c:424 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' This is on debian etch 4.0 asterisk 1.4, it happens quite often everyday and I have to scroll a lot to try to find other error messages. btw can I just put some musica wav files in /var/lib/asterisk/mohmp3 ? that would be great to leave asterisk's processor alone thanks! Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz ___ --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] xten will not send tones to * and i from sip phone
hi there! I have a couple phones connected to a sipura ata and if I go into *- IVR, I press options on the regular phones and it all works fine and dandy. then I connect an xten softphone, a new extension in my dialplan, I dial the ivr, * asks me to dial something to go through it, I press keys on xten, but nothing happens, * just times out through as if I did not press anything! is there some sort of configuration out there to tell the xten softphone to work as expected? thanks! Then another problem! I used the i extension, plus _X and _X. to make sure I catch everything that is not propperly dialed. If I take the regular phones that are connected through the sipura ata, then dial 'exten => 700,1,Goto(default,s,1)' so that I get the asking for an extension to reach, I dial a wrong number and walla, its caight by one of my magic numbers! BUT, if I pickup the same phone, and just dial the same wrong number? I just get a busy signal! and there is nothing registered at the CLI even though I added DEBIG to the configuration! :s What can I do to make sure I always send an error sound and never again a busy signal? thanks! Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front ___ --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] basic 3+ way conference call on plain old phones
hi guys, is it possible to do a basic 3-or-more-way conference call when the phones dont support it? I am fully aware of this concept on expensive phones like this one: Grandstream GXP 2000 -Conference call 3-way http://www.youtube.com/watch?v=hlZ6JqE1MT4 The problem is that the basic plain old commercial PBX supports 3-way calling in ugly old phones like this one: http://www.neo-shop.com/tiendas/0009/varios/telefono%20TEIDE-1.jpg connected to an ata like this one: http://www.egk.com.ar/imagenes/hardware/sipura2.jpg The idea is to be caller (A): dial calle (B), once (B) answers press on HOOK or something else to send them to MOH, then dial callee (C), talk to him a little too, then press the same HOOK or something else and the 3, (A)(B) and (C) in a conference call. Unlike the grandstream, this would definitelly have to be done by *, isnt this part of the basic functionality like voicemail that is already done and a couple lines in the config files it will work on all phones done by *? if not, then, how do you recommend me to it? the closest I have seen to shat I am looking for is http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro is there a better alternative? any thoughts? thanks a lot! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+kids&cs=bz ___ --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] linksys spa3102 for faxing
Hi, I have been considering a purchase of the linksys spa3102 for a couple hours but I would like to know from someone here, wether this device will support faxing on my local asterisk server, I have had success sending and recieving faces with an x100p, and recall that in the old documentation, they mention that if I send/recieve faxes, that it all should be done on the local server for best performanc, so Im asuming tha this device may apply because there will be an ethernet cable between the FXO and the asterisk server? thanks! Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! http://tv.yahoo.com/collections/3658 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
