[Asterisk-Users] Early Media Enable?
Hi, I've searched almost everywhere but have not come across a solution so I was hoping one of your fine folks can help me out. The problem is that a carrier is passing me early media on calls that sometimes have problems connecting. For example, calls to India mobile might play an early media message saying "the phone is out of reach" if mobile is out of area of coverage. Problem is that asterisk does not play this early media message and simply continues to ring indefinitely. Now I know asterisk will not open the audio streams till it gets acks from both sides but is there a way around it? To open one way audio right away? Any solution for this problem? Thanks for any help in advance. Regards, Mohammed Salim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Send DTMF after call bridge
Thanks for replying Alvaro but for me that same dialplan is not working. I
know the digits are sent as the calling card menu doesn't come up and
instead after a few seconds it says "the number you have dialed is invalid".
I have tried putting in multiple w's but like I said, the problem is that
the digits are sent before the call is bridged. I need to send it
afterwards.
Let me know if you have any other solutions please. Thanks.
Regards,
Mohammed Salim
Message: 1
Date: Sat, 24 Sep 2005 23:08:46 +0200
From: "Alvaro G. M." <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Send DTMF after call bridge
To: Asterisk Users
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"
On Saturday 24 September 2005 21:21, Mohammed Salim wrote:
> Hello everyone.
>
> Let me first begin by explaining what I'm trying to do...
>
> I have a calling card that has an access number and requires a PIN to
> be entered and then the number you want to dial, like normal calling
> cards. So what I have done is assign a local DID which when called,
> initiates a Dial to the access number of the calling card.
>
I'm in the same case and you, and the D(digits) options of Dial command
works fine for me. I use it this way:
exten => _9.,1,Dial(SIP/[EMAIL PROTECTED]||D(w6969w${EXTEN}))
Where 6969 is my calling card number (or whatever).
--
Alvaro Gamez Machado.
[EMAIL PROTECTED]
Hazent Systems, S.L.
http://www.hazent.com
C/Rio Caqamares 2, Oficina ocho
28804 Alcala de Henares
Madrid
___
--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] Send DTMF after call bridge
Hello everyone. Let me first begin by explaining what I'm trying to do... I have a calling card that has an access number and requires a PIN to be entered and then the number you want to dial, like normal calling cards. So what I have done is assign a local DID which when called, initiates a Dial to the access number of the calling card. Now, I'm having a hard time figuring out how to send DTMF tones via the dialplan once the call has been bridged. So far I've tried using 'w' in the Dial string to specify the wait period before dialing the digits that follow. I've also tried using the D(digits) option for the Dial application but it clearly says that it will only send the digits once the channel is answered and before it is bridged. So how in the world can you send DTMF via the dialplan to a bridged call? Is it even possible? Thanks in advance for any help. Regards, Mohammed Salim ___ --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 negotiation
The order matters in asterisk so if you want GSM to take priority over G729, simply put that ahead of the G729... so your settings should be: Allow=all Allow=gsm Allow=g729 Allow=ulaw Allow=alaw Try that and see if it works. Regards, Mohammed Salim EZZI Telecom, Inc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Eissler Sent: Tuesday, January 25, 2005 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Codec negotiation The codec is selected by asterisk depending upon the codecs that you have allowed for the particular channel context and your setting of the bandwidth= parameter. It would be nice if you could set things up so that an inbound call could force * to a higher bandwidth codec when needed (for example, an inbound fax call, let's say) but AFAIK this is not possible. -mark On Jan 25, 2005, at 10:18 AM, <[EMAIL PROTECTED]> wrote: > > Hello > > On every Incoming SIP and IAX call I see the following in asterisk > debug: > > Accepting AUTHENTICATED call from 10.10.10.10, requested format = gsm, > requested prefs = (), actual format = g729, my prefs = > (g729|gsm|g723|g726|ulaw|alaw) priority = mine > > The problem is that the codec preference on both parties is different > > The calling party has preference gsm/g729/etc > The called party (the one you see this debug from) has preference > g729/gsm/etc > > The problem is.. This call is now set up with G729... And I want it > rather to be decided by the callING party (thus want the call to be > negotiated GSM) > > What can I do about this? (I just want that if I receive a call the > calling party decides the codec, and not my side) > > My IAX.conf and SIP.conf have the following allow settings now > > Allow=all > Allow=g729 > Allow=gsm > Allow=ulaw > Allow=alaw > > > Help :-) > > > ___ > 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 > -- 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] chan_sip.c:7325
Hi, I've searched everywhere but I have not been able to get an answer to the following problem: I get the following notice (appropriate parts are taken out for security purposes) after long periods of registration using an spa 2000 with registration set to 3600 seconds and a proxy failover set to 20 seconds. I use version 2.08 as it has stayed registered longer than the current version 2.10(e). Also, I have tried a few spa2000's so I know it's not related to any particular box that has gone bad. Oct 31 00:01:01 NOTICE[-1116333136]: chan_sip.c:7325 handle_request: Registration from ' @>' failed for '' The box stays registered for a random period of time before resulting in the above message. It remains unregistered for another random period of time before becoming registered again. And the cycle continues. Any help would be very much appreciated. Regards, Mohammed Salim EZZI Telecom, Inc. ___ 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] sipura registration problem
Yes my registration is also set to 60 sec. I'm going to upgrade them to 2.10 and see what happens. Thanks for your reply demitri. Regards, Mohammed Salim EZZI Telecom, Inc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of reseaux Sent: Tuesday, September 21, 2004 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sipura registration problem Dear Mohammed I have notice the same problem of UNREGISTRATION of my Sipura 2000 i have some of this with R2.07 and other with 2.09 i also try to put registration expire to 60 sec but seems the same i try to made more debug on this.. Bye Dimitri On Tuesday 21 September 2004 17:17, Mohammed Salim wrote: > Hi everyone, > > > > I'm having an odd problem with one of my sipura boxes. The box registers > the first time with asterisk properly after being plugged in. After which, > some of the subsequent registration tries fail and the box becomes > unregistered. However, after a few hours, it finally successfully > re-registers and the cycle continues. I have not been able to figure out > the problem but I'm running the same config for this sipura as I am for > about 5 others. I have not ruled out the possibility that the box might > actually be bad. I am running software version 2.07e on all of the sipuras > and this is the only one that is causing problems. > > > > Any suggestions would be appreciated. > > > > Regards, > > Mohammed Salim > > EZZI Telecom, Inc. ___ 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] sipura registration problem
Hi everyone, I’m having an odd problem with one of my sipura boxes. The box registers the first time with asterisk properly after being plugged in. After which, some of the subsequent registration tries fail and the box becomes unregistered. However, after a few hours, it finally successfully re-registers and the cycle continues. I have not been able to figure out the problem but I’m running the same config for this sipura as I am for about 5 others. I have not ruled out the possibility that the box might actually be bad. I am running software version 2.07e on all of the sipuras and this is the only one that is causing problems. Any suggestions would be appreciated. Regards, Mohammed Salim EZZI Telecom, Inc. ___ 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] BroadVOX
Hi Guys, I work with Dan and yes this whole Broadvox problem is a huge piece of [EMAIL PROTECTED] However, I've noticed that there is a useragent= cmd that can be used in the sip.conf. Right now it's set to useragent="Asterisk PBX". I was wondering if we change that to reflect a Cisco Gateway or something, will that mean that Broadvox won't realize it's an asterisk PBX system. In other words, how exactly does the useragent= cmd work? If we can fool them into thinking it's not asterisk, then they won't have a problem. They claim that asterisk doesn't send correctly formatted sip messages which is completely wrong! They are in cisco's back pocket is what I think. In any case, can someone please tell me if what I'm trying to do is possible? Thanks, Mohammed Salim EZZI Telecom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Mahoney, System Admin Sent: Tuesday, August 17, 2004 12:53 PM To: Asterisk Mailing List Subject: [Asterisk-Users] BroadVOX Guys, For what it's worth, after months of trying to troubleshoot issues with them, and after paying them around $2500 for setup and a down payment (it's unclear what of that will be refunded, if any) BroadVox -- http://www.broadvox.net/ -- decided to terminate our contract without any valid reason, and the only explanation they could cite was "it's because of the software you're using". We use asterisk. We've asked about using other products (such as SER), and they say they don't want to accept *anything* open-source. Apparently they're a cisco shop, and maybe their TAC contract only supports getting calls from other cisco devices (but this is conjecture). I wanted to send this email for two reasons. First, to warn the community to stay away from them in general. More than once we've had issues where we were sure our configuration was dead-on, but Broadvox would be, for example, delivering different inbound DID's with different DTMF encodings, after telling us up and down that everything was right. Their "support" person, Alex, has an attitude the size of Montana, and is easily offended if you can't understand him (through a thick russian accent) on the phone. He refuses to communicate via email, and if you give him an attitude, he'll simply threaten you. Secondly, the ONLY good thing about Broadvox, was the rates they were offering us. Without getting into a debate here (please feel free to forward me recommendations off-list), does anyone know a good provider who is willing to provide nationwide DIDs at a reasonable cost? Thanks for all the help, -Dan Mahoney -- Dan Mahoney Techie, Sysadmin, WebGeek Gushi on efnet/undernet IRC ICQ: 13735144 AIM: LarpGM Site: http://www.gushi.org --- ___ 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] DID Questions
Which device did you assign the 877 DID to? Because if you call that device using another device registered to your asterisk box (i.e. your xlite softphone) then its not hitting your provider at all. The asterisk sees the call come from a registered sip device and finds that there is a registered device that has the assigned DID and routes the call to it, completing it. However, when you call using PSTN, that's when the provider has to route the call to your asterisk pbx. And it seems to me, they aren't doing that properly. Unless I'm totally not understanding what your trying to say but I've been in these situations many times and its usually the provider's fault. Mohammed Salim EZZI Telecom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Roberts Sent: Monday, August 16, 2004 6:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DID Questions I'm using SIP and I'm not getting anything in the CLI when I call over the PSTN. But over my Sip Softphone I can call it, no problem. ___ 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] DID Questions
You should see the call come in even if its not part of your dial plan. The call would default to your default context if no match is found. But either way, the call would be seen in your asterisk CLI even with the sip debug turned off. This might seem like a stupid question but just to be sure, you are receiving this call via ip, correct? Mohammed Salim EZZI Telecom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Roberts Sent: Monday, August 16, 2004 5:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DID Questions Is there anyway to test if this call is touching my servers? Everyone is telling me the DID is fine. But I can't confirm that for sure. And I don't want to go ahead and start making changes to my config when the DID isn't even working in the first place. ___ 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] DID Questions
Seems like your provider hasn't configured the DID properly and its not even hitting your system. Check with them. Mohammed Salim EZZI Telecom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Roberts Sent: Monday, August 16, 2004 4:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] DID Questions I'll try to keep this short and sweet, I don't want to confuse you, or myself.. So here goes I had a DID, it went down, I got a new DID (800#) and I changed the exten=>###-###- to the new 800 number in extensions.conf I'd done lots of changes to the config files and nothing seems to be doing anything, When I dail the 800 number all I get is a fast busy signal. When I log into the CLI, it shows me nothing when I dail the number. It used to show me pages and pages of info when someone dialed in. I'm new to asterisk/voip and I'm lost. ___ 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
