[asterisk-users] PJSIP Bind Issue

Hi, I've got a test server with two IPs (one IP is virtual and moves to a backup server if the first goes down). I'm trying out PJSIP and specified the virtual IP for the Bind address of all transports I'm using. All SIP packets are now sent of the virtual IP. But the RTP UDP data is being sen

[asterisk-users] Voicemail ODBC Storage

Hi, Is there any reason why ODBC voicemail storage requires varchar for most fields? For example, is there anything stopping me using a BIGINT or similar for origtime or INT for duration? Kind regards, Dan -- _ -- Bandwi

Re: [asterisk-users] auto-answer call

Ø when i use the Dial the sip/105 still ringing This should help you out http://wiki.snom.com/FAQ/How_to_make_Asterisk_send_INVITEs_to_trigger_the_phone_for_Intercom Dan Journo Kesher Communications (UK) www.keshercommunications.com<http://www.keshercommunications.

[asterisk-users] Gigaset R630H and Asterisk

hold, but the caller can hear the agent. Any ideas? Thanks Dan Journo Kesher Communications (UK) www.keshercommunications.com<http://www.keshercommunications.com> -- _ -- Bandwidth and Colocation Provided by http://w

Re: [asterisk-users] Call Queue advise

>yes but I believe that least recent would ring one agent at a time? If my >understanding is incorrect please correct it. We are wanting to keep with >multiple phones ring to ensure coverage. >From what I've seen, I don't think this is possible. But maybe ask in the >#asterisk channel on Free

Re: [asterisk-users] Asterisk SIP server on windows

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi Sent: 04 December 2013 09:08 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Asterisk SIP server on windows Hi all, I need to build an

Re: [asterisk-users] userfield not logged to CDR

Ø Ok, Ive done a little more digging and it appears that some lines in CDR-CSV are also being missed out intermittently. Ø But the userfield in CDR-CSV is populated every time. Ø Its Asterisk 11.5.1 Ø Ignoring ODBC for now, how do I debug writing of CDR-CSV? I've logged this as an issue

Re: [asterisk-users] userfield not logged to CDR

Ø I'm logging cdr via odbc to mysql. It seems that there is an intermittent problem where the CDR(userfield) isn't written to the database. The rows all seem to be there, but that specific field is missing. The same dialplan. Nothings changed. Probably 1 in 10 is missing the userfield. Ø Any

[asterisk-users] userfield not logged to CDR

Hi, I'm logging cdr via odbc to mysql. It seems that there is an intermittent problem where the CDR(userfield) isn't written to the database. The rows all seem to be there, but that specific field is missing. The same dialplan. Nothings changed. Probably 1 in 10 is missing the userfield. Any i

Re: [asterisk-users] Realtime vs Static Files

> It is really unbelievable ... I was thinking: Asterisk uses an internal > database to maintain states of peers. It is usually located in > /var/lib/asterisk/astdb and it is a berkely db, but other database backends > seem available. Are you sharing also this database between the two servers?

Re: [asterisk-users] Realtime vs Static Files

>> Upgrading to the latest version didn't help. After about 30 minutes, >> Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as >> Registered on Asterisk1. > It is something really amazing... Can you run "sip show peers" on each one of > the servers and post the response? >

Re: [asterisk-users] Realtime vs Static Files

> I am curious, is your version of asterisk correctly compiling the regserver > field? Each server needs to have a distinct server name. Upgrading to the latest version didn't help. After about 30 minutes, Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as Registered on As

Re: [asterisk-users] Realtime vs Static Files

>> Its probably an issue with the version of Asterisk we are using because I >> haven't had this problem in the past. > I am running the latest 1.8 version. Which version are you running? 1.8.15.0. I'll upgrade it to 1.8.20.1 when I can and see if it makes a difference. -- _

Re: [asterisk-users] Realtime vs Static Files

> Maybe you are lacking some of the configuration. These is the relevant part. > rtcachefriends=yes > rtsavesysname=yes > rtupdate=yes > rtautoclear=yes We have rtcachefriends=yes rtsavesysname=yes and these we don't have but they are set to YES by default > rtupdate=yes > rtautoclear=yes Its p

Re: [asterisk-users] Realtime vs Static Files

> We have never experienced that and use realtime with multiple asterisk > servers. We've only recently started seeing the problem. To simplify the issue, assuming we have two servers, Asterisk1 and Asterisk2... Asterisk1 is a primary server and Asterisk2 is a backup and used as a failover. As

Re: [asterisk-users] Realtime vs Static Files

ot;sip reload" when needed, I can get the same functionality as realtime without the issues described above. Have I missed something? Thanks Dan -- Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<http://www.k

[asterisk-users] Realtime vs Static Files

Hi, We're trying to decide whether to switch back to a static file for sip.conf. Currently we use mysql realtime but can't see any real benefit. Why would someone choose realtime sip over static files? Thanks Dan Journo Kesher Communications (UK) Business Phone Sy

[asterisk-users] Virgin Meda VMDG280 and SIP Asterisk

options on the webadmin of the router. Has anyone successfully connected a SIP endpoint (Polycom IP331) with Asterisk on Virgin Media? Kind regards, Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<http://www.

[asterisk-users] Connecting to an Old Phone System

0e Many thanks Dan Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<http://www.keshercommunications.com/hostedpbx.html> -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Answering machine answers after pickup a phone.

ler will hear silence because Asterisk will pick up the phone as soon as it starts ringing. In addition, if the caller happens to be talking before the call is answered, BackGroundDetect(silence/10) won't work. Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.kesher

Re: [asterisk-users] pickupgroup

Thanks Paul. That link looks great. Is there more information available on how to apply patches? Im not sure how patches work. Do I have to use a specific version such as 1.4.20, or can I use any of the 1.4.x branch? Thanks Dan --

Re: [asterisk-users] pickupgroup

Is there any technical reason for this 63 group limit? >>>Currently the number of callgroups/pickupgroups is limited to 63, because >>>the variable type that holds the group bits is a long long. >> Ah, so it's a major change to get it to hold a larger number? >> I was hoping I could just c

Re: [asterisk-users] pickupgroup

>> Is there any technical reason for this 63 group limit? >Currently the number of callgroups/pickupgroups is limited to 63, because the >variable type that holds the group bits is a long long. Ah, so it's a major change to get it to hold a larger number? I was hoping I could just change the fo

Re: [asterisk-users] pickupgroup

> My take on this is that asterisk just isn't multi-tenant friendly. Is there any technical reason for this 63 group limit? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for

Re: [asterisk-users] pickupgroup

> Why don't you just use an AGI/Table to control the pickup group? You can use > contexts or other granular features to keep any number of clients in this 63 > count (IMO). Thanks for your reply. How would I use an AGI to handle the pickup request? Would I control the whole pickup request? Or

[asterisk-users] pickupgroup

Hi, Using 1.4, I see that pickupgroup can only be between 1 and 63. We run a hosted PBX service and need to give our client access to the call pickup feature. I thought that I could simply use the client's ID number for the pickupgroup number. The client's ID number is generated by our CRM sof

Re: [asterisk-users] MixMonitor and attended transfers

ing, even when you transfer the call. You may need to add /n http://www.voip-info.org/wiki/view/Asterisk+local+channels Hope that helps. It's a little hard to explain, but try it out. Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercommu

Re: [asterisk-users] check_auth: username mismatch

> I've got a Polycom 501 that I just can't seem to get lines 2 and 3 to work > on. Line 1 works fine. Last time I had that issue, it resolved itself when i restarted Asterisk. Are you able to do that? Regards Dan -- _ -- Band

Re: [asterisk-users] SIP Peer Name Variable

> pbx*CLI> core show function CHANNEL > -= Info about function 'CHANNEL' =- Thanks, but i'm using 1.4. The most info I get is this:- -= Info about function 'CHANNEL' =- [Syntax] CHANNEL(item) [Synopsis] Gets/sets various pieces of information about the channel. [Description] Gets/set vario

Re: [asterisk-users] Cannot figure out pound key in qwerty keyboard

> I am new in Asterisk. I am using asterisks with freepbx 2.7.0.10 version. I > have tried to setup voice mail by dialing *97 from my extension. The > prerecorded system asking for a "pond key" at the end of each recording. But > unfortunately i am not able to locate a pound key on my qwerty ke

[asterisk-users] atxfer fails to read data

Hi, We are having a problem that is preventing users from using *2 to manage an attended transfer. After dialling *2, asterisk places the call on hold, but you can only dial one digit before it times out, and the cli says:- [2011-06-29 18:33:55] WARNING[29877]: res_features.c:938 builtin_atxfer

Re: [asterisk-users] Connect intercom to Asterisk?

er done it myself, so I can't recommend a suitable intercom. Hopefully someone else can. Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<http://www.keshercommunications.com/hostedpbx.html> -- __

Re: [asterisk-users] Call recording - methodology

> I am at a loss. Can you pastebin the following:- - Run asterisk-cvvvddd and paste the output - Pastebin your features.conf - Pastebin your extensions.conf I'll see if I can spot anything obvious. Dan Journo Kesher Communications (UK) Business Phone Syst

Re: [asterisk-users] Call recording - methodology

gh '1' on > SIP/6000-002e It looks like Asterisk hasnt added the new details from features.conf. You may need to fully restart Asterisk in order to get this to work. Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercommunications.com/> | Hosted

Re: [asterisk-users] Call recording - methodology

eceiving the tones correctly. What method are you using to transmit the dtmf tones? Regards Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<http://www.keshercommunicatio

Re: [asterisk-users] Call recording - methodology

C_FEATURES=MixMonApp, either declared in your globals section of extensions.conf, or used in a Set(DYNAMIC_FEATURES=MixMonApp) fashion on a per channel basis in extensions.conf. Thanks to Warren Selby from http://www.selbytech.com for pointing that out. Dan Journo Kesher Communications (UK) Busines

Re: [asterisk-users] Call Recording using MixMonitor - close, but would like some more words of wisdom.

> DYNAMIC_FEATURES=MixMonApp, either declared in your globals section of > extensions.conf, or used in a Set(DYNAMIC_FEATURES=MixMonApp) fashion on a > per channel basis in extensions.conf. Sorry, i forgot to mention that one. Dan Journo Kesher Communications (UK) Business Phone Sys

Re: [asterisk-users] asterisk login to voicemail

sterisk to pass variables along with the call. Otherwise, if you transfer the calls within the dialplan, the variables you set get left behind. Hope that helps Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercomm

Re: [asterisk-users] asterisk login to voicemail

debug and make sure ${MAILBOXID}@${MAILBOXCONTEXT} is correct and the mailbox exists. Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<http://www

Re: [asterisk-users] asterisk login to voicemail

,Playback(astcc-please-enter-your) exten => a,n,VoicemailMain(${MAILBOXID}@${MAILBOXCONTEXT}) When the user presses *, they are passed to the 'a' extension above and into VoicemailMain. I'm sure you can turn this into AGI easily enough if needed. Dan Jour

Re: [asterisk-users] Call recording - methodology

to recording every call, we add this before the DIAL(SIP/extension) command in extensions.conf:- exten => _9.,14,Macro(mixmon,nobeep) If you don't want to record every call, you can give the operator the option of press *1. We did this by adding the following to features.conf:-

Re: [asterisk-users] Status of Queue Members

Probably this will help you... http://ofps.oreilly.com/titles/9780596517342/ch13.html#ACD_id288901 Check the section 'Controlling when to join and leave a queue'. Thanks. Thats perfect! Kesher Communications (UK) Business Phone Systems | Hosted PBX

[asterisk-users] Status of Queue Members

Hi, I'm trying to work out an issue with call queues. I need the calls that are in a queue to be kicked out if all members are unavailable (for example if all SIP members are having network problems). I tried leavewhenempty = yes but that only seems works when all queue members specifically lo

Re: [asterisk-users] HK DIDs

Sorry, just realised I posted this to the wrong mailing list. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

[asterisk-users] HK DIDs

Hi, Does anyone know a source for multi-channel Hong Kong DIDs (probably 10 channels to start with) that can receive international calls? We are being told that multi-channel DIDs can only receive calls from within Hong Kong. Thanks Dan Journo Kesher Communications (UK) Business Phone

Re: [asterisk-users] [Opinion Request] SIP phones that work well with Asterisk

phone, and the phone can be rebooted through Asterisk to update the settings instantly. It also stores the phone's directory on the FTP server so users don't lose any contacts if the phone needs replacing. Hope that helps. Dan Journo Kesher Communications (UK) Business Ph

Re: [asterisk-users] doorphone?

way like the PAP2T. Otherwise, you need a network connection directly into the doorphone unit, and some people don't like that because it can give a hacker/burglar, direct access to your internal network. Hope that helps. Dan Journo Kesher Communications (UK) Bus

Re: [asterisk-users] DTMF and Snom

ou always had the problem? Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<http://www.keshercommunications.com/hostedpbx.html> -- _ -- Bandwidth an

Re: [asterisk-users] Asterisk pass a call to status answer while still ringing

To unsubscribe, go to this address: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

[asterisk-users] Problem When Using Polycom with 2 Lines

Hi, Has anyone had a problem setting up two registrations (on the same Asterisk server) on one Polycom phone? When the user tries to make a call on the 2nd line, it works fine. But when they try the first line, the CLI says:- Using INVITE request as basis request - 9f5fe9a5-215d0f3a-b2fbe...@19

[asterisk-users] Nat Issue - I think

Hi, I'm using qualify= on my asterisk server that provides outgoing pstn calls to a few companies. I've got one client in particular that has their own asterisk server which is connected to my server. This client seems to be having a nat issue. It's not a connectivity issue as i've tried cons

Re: [asterisk-users] Why are the hackers scanning for these?

> Here's some examples: >2648061411 >3190339404 I'm getting exactly the same. Odds of getting a working number, are like the odds of winning the lottery. My guess is they are either trying to find a voip trunk, or they are trying to make cold calls to the extensions on my system. Sales or somet

Re: [asterisk-users] One way voice with Asterisk

> When I dial into Asterisk ( I have a SIP trunk - which I need to make > sure is not faulty), I only get one-way voice communication. > The calling party, from the SIP trunk hears nothing - the extension > rings on the Asterisk server (you can see it in the CLI and hear it at > the computer), a

Re: [asterisk-users] ADSL Load Balancing

The adsl lines are with separate providers, so that won't work. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http:/

Re: [asterisk-users] ADSL Load Balancing

> I use the SG560 (http://www.snapgear.com/index.cfm?skey=1557) to do this. Thats perfect. Any idea where they are available? I cant locate a store online. Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.ap

[asterisk-users] ADSL Load Balancing

Hi, I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connectio

Re: [asterisk-users] billsec=0 when using Local channel

Hi, I've got a dialplan that transfers all outgoing calls to a Local channel before dialling out via SIP. I did this because sometimes i'm dialling two numbers at the same time and need to know which call is answered for billing purposes. However, I've just noticed that billsec is always equal

[asterisk-users] billsec=0 when using Local channel

Hi, I've got a dialplan that transfers all outgoing calls to a Local channel before dialling out via SIP. I did this because sometimes i'm dialling two numbers at the same time and need to know which call is answered for billing purposes. However, I've just noticed that billsec is always equal

Re: [asterisk-users] Extension Exists

Thanks Leif, Forgot I could do a db lookup for the ddi. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://w

[asterisk-users] Extension Exists

Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551...@incoming_calls) Currently, I'm paying for all calls, regardles

Re: [asterisk-users] Email from Dialplan

thanks. Ill give that a go. Steve Edwards wrote: On Wed, 20 Oct 2010, Dan Journo wrote: > I want to set up a warning so that after a Dial cmd, if the DIALSTATUS is > CHANUNAVAIL, Asterisk sends an email to the > admin to check the voip phone is connected properly. > > I'

[asterisk-users] Email from Dialplan

Hi, I'm sure this topic has been discussed before but i'm having trouble finding a simple answer. Whats the easiest way of sending an email from Asterisk? I want to set up a warning so that after a Dial cmd, if the DIALSTATUS is CHANUNAVAIL, Asterisk sends an email to the admin to check the vo

Re: [asterisk-users] Recording

Thanks. Thats perfect! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users m

Re: [asterisk-users] Recording

Non-Commercial Discussion' Subject: Re: [asterisk-users] Recording From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Monday, October 18, 2010 2:37 PM To: Asterisk Users Mailing L

[asterisk-users] Recording

Hi, Does anyone have a professional recording of someone saying "Recording" so I can let the operator know that the one-touch recording has started successfully? Thanks Dan [cid:image001.gif@01CB6F04.2EEFF060] Dan Journo IT Business Consult

Re: [asterisk-users] DMTF Mode

> I recommend reviewing [1] and look for possible regressions. There have been some update to RFC2833 over the last few months. In 1.4.36, its a regression issue. I switched to 1.4.35 and no problems since. Can anyone else confirm this for me? Problems are either no dtmf tones being processed by

Re: [asterisk-users] DMTF Mode

s Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: 17 October 2010 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode > Have you tried rela

Re: [asterisk-users] DMTF Mode

> Have you tried relaxdtmf or rfcc2833compensate? Just tried it, but it didnt make a difference. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

Re: [asterisk-users] DMTF Mode

Whats payload used for in rfc2833? I'm wondering if that is incompatible. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: 17 October 2010 14:54 To: Asterisk Users Mailing List

Re: [asterisk-users] DMTF Mode

I'm really struggling with this DTMF issue. In order to test it, I've tried a few different providers and DTMF RFC2833 does work with any of them, even though a few of them insist that it is. Is this a bug with 1.4.36? Has anyone else experienced this problem? The Asterisk CLI is showing the DT

Re: [asterisk-users] Remote Unix Connection

> Some service is definitely connecting to your asterisk using AMI. Such > services use username/password described in manager.conf. Usually its is some > monitoring service. Although the message says 'remote UNIX connection' but it > can be very well something from localhost. I would suggest to

Re: [asterisk-users] Remote Unix Connection

Nope, Its a totally normal self-built Asterisk. Dan Zeeshan Zakaria wrote: Do you use FreePBX by any chance? Zeeshan A Zakaria -- www.ilovetovoip.com<http://www.ilovetovoip.com> On 2010-10-16 6:38 PM, "Dan Journo" mailto:d...@keshercommunications.com>> wrote: >

Re: [asterisk-users] DMTF Mode

Hi All, Regarding the DTMF issue I reported where the tones werent being sent through the provider to the pstn phones, I ended up being told to switch to "inband". However, now, asterisk is not recognising my features (*1, etc). Any ideas? I've checked using tcpdump, and asterisk is still par

Re: [asterisk-users] Remote Unix Connection

> Serious answer: > Looks like a process running asterisk -r. Do you have any sort of > AGI, cron job or perhaps a nagios check which does this? > Not so serious answer: > IT IS COMING FROM INSIDE OF THE HOUSE Thanks for lightning my day! Is there any way to debug this because as far as i'm awa

[asterisk-users] Remote Unix Connection

Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. so

[asterisk-users] (no subject)

Hi, Does anyone know where this is suddenly coming from? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.

Re: [asterisk-users] DMTF Mode

> Since alaw is "sort of making it work", either your SIP provider or your other party is not in U.S. Its stopped working again. This is really unusual. I didnt change anything. I decided to do a tcpdump, and I can clearly see the rfc2833 packets being exchanged correctly. Why should both partie

Re: [asterisk-users] DMTF Mode

Thanks to everyone who helped me on this. Hopefully the provider can sort out the sticking tones now. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] DMTF Mode

> According to the WIKI, changing rfc2833 to auto in sip.conf should do the trick. Didnt help. I'm contacting the provider to see if they have any ideas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] DMTF Mode

> From what I read, the codec could be trying to switch from rfc2833 to inband during the call, causing the "stuck" effect. Any way to prevent that? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] DMTF Mode

> Since alaw is "sort of making it work", either your SIP provider or your other party is not in U.S. I dont understand why the codec should make a difference if im using rfc2833. Could you clear that up for me? -- _ -- Bandwid

Re: [asterisk-users] DMTF Mode

> Since alaw is "sort of making it work", either your SIP provider or your other party is not in U.S. They arent in the US. Everything is in the UK. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] DMTF Mode

> I just added alaw to the provider's list, so now it reads (gsm, ulaw, alaw) > and its started working in a fashion. > The DTMF tones keep "getting stuck". I press a number on the sip phone, and > the other party hears a tone. But every few tones, it gets stuck and they > hear a long tone of a

Re: [asterisk-users] DMTF Mode

> What is your featuredigittimeout value? Not used. So default 1000ms. I just added alaw to the provider's list, so now it reads (gsm, ulaw, alaw) and its started working in a fashion. The DTMF tones keep "getting stuck". I press a number on the sip phone, and the other party hears a tone. But

Re: [asterisk-users] DMTF Mode

I'm on 1.4.30 and this is what I get using debug 5 -- Accepting AUTHENTICATED call from 192.168.xx.xx: > requested format = ulaw, > requested prefs = (ulaw|gsm|alaw), > actual format = gsm, > host prefs = (slin|gsm|ulaw|alaw), > priority = mine Strange. I dont

Re: [asterisk-users] DMTF Mode

Could features.conf be preventing asterisk from repeating the DTMF tones? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] DMTF Mode

> > How can I tell if Asterisk is sending the tones through to the provider? > You need to enable DTMF logging (logger.conf) and debug an incoming / outgoing call. Can you understand this? I can see the DTMF signals coming in. I pressed 5 on the normal phone line, and then I pressed 8 on the s

Re: [asterisk-users] DMTF Mode

> Based on this, your call is probably getting to the provider as ulaw (the > alaw is thrown out since it isn't in both selections; if you are in U.S. you > don't need the alaw). Try the call with higher debug (at least 5) and verify > which one is being selected. debug 5 doesnt give me any in

Re: [asterisk-users] DMTF Mode

> I would suggest first to make sure that asterisk is receiving DTMF fine from > your IP devices/phones. Do you have a test IVR where you can dial and press > digits and verify that asterisk is responding? Made a quick IVR, and its working for both sides of the asterisk (between the provider an

Re: [asterisk-users] DMTF Mode

> Are you using the .gsm codec or some other flavor (ulaw, alaw, G729?) This is from the sip.conf for the provider: allow=gsm allow=ulaw This is from the sip extension:- alaw,ulaw,gsm -- _ -- Bandwidth and Colocation Provided b

Re: [asterisk-users] DMTF Mode

> Can you send us the SIP config of the sip provider (in sip.conf), removing > appropriate passwords and static IPs of course. [provider] type=friend host=removed username=removed fromuser=removed secret=password context=incoming_calls dtmfmode=rfc2833 < also tried "auto". disallow=all allow=gsm

Re: [asterisk-users] DMTF Mode

> Then that likely means your phone have the correct dtmfmode, but the link > between you and the provider doesn't. Just carried out another test to see if my provider was working properly:- exten => INCOMINGDDI,1,Wait(1) exten => INCOMINGDDI,n,Answer() exten => INCOMINGDDI,n,SendDTMF(12345) If

Re: [asterisk-users] DMTF Mode

I just tried this:- [test_calls] exten => 555,1,Answer() exten => 555,n,SendDTMF(12345) exten => 555,n,Playback(beep) I dialed 555 on the sip phone, nothing was heard, and then a beep... It seems that Asterisk isn't sending DTMF. Its only able to receive. Thanks Dan -- ___

Re: [asterisk-users] DMTF Mode

> It depends upon whether you are receiving DTMF or sending, and whether you > are using a VoIP protocol or using DAHDI/Zaptel. Sorry about the lack of info. It's a simple SIP only setup. A handful of sip phones, an asterisk server, and a sip provider. The DTMF signals from the sip phones are

[asterisk-users] DMTF Mode

Hi, Which DTMF mode do people mostly use? I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for feature usage), the tones arent repeated to the end user. So if I call a company that has a menu system, I can't use the menu. Thanks Dan -- ___

[asterisk-users] Application Map Not Working

Hi, I'm using the applicationmap in features.conf to allow the user to press *1 and run a macro which records using MixMonitor. All was fine in our office (behind a nat). But when I took the sip phones to an end user (also behind a nat), I found that *1 didnt work for outgoing calls. But for s

Re: [asterisk-users] MWI Assistance

> I'm struggling to get the MWI set up on a few Polycom phones. Sorted. From voip-info. http://www.voip-info.org/wiki/view/Asterisk+RealTime The database peers/users are not kept in memory. These are only loaded when we have a call and then deleted, so there's no support for NAT keep-alives (q

[asterisk-users] MWI Assistance

Hi, I'm struggling to get the MWI set up on a few Polycom phones. The setup is like this. I've got a few phones in the context called [company2_phones] and I've got a few mailboxes in the voicemail context [company2]. Therefore, for each entry in sip.conf (i'm actually using sip realtime if th

Re: [asterisk-users] Flash WAV Player

> Does anyone know of a Flash or Java player that can play WAV files created by > Asterisk? Found one. http://blog.datacompboy.ru/2010/01/27/wavplayer-1-7-1-full-js-api-and-support-for-reversed-order-bits-lu-and-la/ -- _ -- Ba

[asterisk-users] Flash WAV Player

Hello, Does anyone know of a Flash or Java player that can play WAV files created by Asterisk? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introduct

  1   2   3   >