Re: [Asterisk-Users] IAX only speech one way

Thanks for your suggestion. Unfortunately, it didnt change anything, A can still not hear B, but B can hear A, strange.. Michael 2005/10/19, Rich Adamson <[EMAIL PROTECTED]>: > > > I have two Asterisk's connected via IAX, they are sitting on the same > > network, via a VPN, so there should b

[Asterisk-Users] Asterisk hangs

Since some CVS Updates the asterisk hangs after command: reload or restart now. Then i have to kill -9 th eprocess. Nothing will be outout inside the CLI but i can type commands. Somebody know this problem? And the CallerID bug still seems to be in there too. Regards rene _

Re: SV: [Asterisk-Users] Queues and call waiting indication

On Tue, 2005-10-18 at 14:35 +0200, [EMAIL PROTECTED] wrote: > Hi, > > This issue has been discussed probably a million times on every asterisk > forum in the world and I have the same problem too. Another problem you would > have with the agents is that when they make an outgoing call they are

[Asterisk-Users] Problems Calling PSTN PSTN FROM ASTERISK

Hi I terminated a call through SIP to a landphone i have the following problems. 1.) asterisk gives a fake riming tone, it does not give the real tone from the phone company. 2.) when I put the call on hold the on hold music is not very clear. but when I talk the call quality is very clear.

SV: SV: [Asterisk-Users] Queues and call waiting indication

Could you post an example of how you've solved it. I read something about this earlier but didn't quite figure it out. I already use AgentCallbackLogin... And I still don't understand why this behavior isn't standard for queues. Does this really fix the "agent makes an outgoing call but still re

Re: [Asterisk-Users] IAX only speech one way

Mir wrote: > Thanks for your suggestion. > > Unfortunately, it didnt change anything, A can still not hear B, but B > can hear A, strange.. > I had the same problem with one of my IAX providers in AUS. Both ends turned of trunking and all was fine with the world again. Not sure what was th

[Asterisk-Users] Realtime - table voicemail

I use E.164 number as customer-id and mailbox-id. E.164 can consist of 4 digets of country code, 4 for area and 4 for the switch and 4 for the user, which gives you a total length of 16 digits. How can I modify the table voicemail to allow me that, and is a change of the table enough? bye

RE: [Asterisk-Users] Voicemail as an email attachement

I changed my app_voicemail.c to work not with sendmail but with sendEmail that connects to any SMTP and sends email with attachment... It's dirty, but it works. If you are interested I can upload app_voicemail.c and sendEmail package somewhere.. >I have configured the voicemail.conf file as per

[Asterisk-Users] SER and Asterisk

I have on one machine Openser and Asterisk. Since Asterisk was first, I let it have the port 5060 ;-) I have choosen for Openser the port 5062. I tried several hard and soft phones to connect to ser to the port 5062, however each of the phones tries to connect to asterisk. I am totally confu

Re: [Asterisk-Users] Audiocodes MP-108

Hello Jeremy, I have been using the MP-108's with H323 interface in a project over one year ago and I found them to be quite good and easily interoperable. After a while both units seemed to lose the IP address when turned off, while retaining other parameters, so it's quite a nuisanmce, b

Re: [Asterisk-Users] Asterisk hangs

Hi Rene, Yes, I've seen that but our version from CVS is a month or so old os it may well have been rectified now. On our version reloads cause the process to die about 50% of the time, work fine about 45% and cause it to hang in the way your describe probably 5%. Simon On 19/10/05, René Enskat [

Re: [Asterisk-Users] SER and Asterisk

hello,  trace the SIP packets and see if they are actually addressed to 5062. if you post the ngrep or ethereal dump we'll see whats actually going on. I do this with SER on 5060 and asterisk on 5070 and there are no problems -  my extensions point to 5060 and my DID's point to 5070 so asterisk ser

[Asterisk-Users] TDMoE question

I am asked to consider deploying asterisk servers as soft-switches on a large scale, but wanted to preserve TDM properties of a call, especially for modem applications which some of the end users may want. I was thinking TDMoE may work well for this, at least on the surafce but had specific questi

Re: [Asterisk-Users] DID setup from goiax.com

On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote: > Can anybody post a step by step setup guide, please? Its like anything else once you have signed up ... in iax.conf register => PHONENUMBER:[EMAIL PROTECTED]/goiax-in [goiax] type= peer host= server1.goiax.com c

Re: [Asterisk-Users] SER and Asterisk

On Wed, 2005-10-19 at 10:55 +0200, Yair Hakak wrote: > hello, > trace the SIP packets and see if they are actually addressed to 5062. > if you post the ngrep or ethereal dump we'll see whats actually going > on. I do this with SER on 5060 and asterisk on 5070 and there are no > problems - my exte

[Asterisk-Users] How to suppres leading zeros in zapata.conf?

Hi, my asterisk is running behind a Siemens HiCom, connected via ISDN. Connection to that Hicom equipment is madewith an AVM Fritz! card. I use a HFC card for the local trunk (NT mode with zaphfc). My problem: something adds an additional leading 0 to all inbound calls (except those coming fr

[Asterisk-Users] Persistant connection for MYSQL command

When doing mysql commands, such as: exten => _X.,1,MYSQL(Connect connid localhost dbuser dbpass dbname) exten => _X.,2,MYSQL(Query resultid ${connid} SELECT\ scriptname\ from\ mac2pin\ where\ userid=${CALLERIDNAME}) exten => _X.,3,MYSQL(Fetch fetchid ${resultid} AGIScript) exten => _X.,4,GotoIf

[Asterisk-Users] Problem with select correct network interface (oh323)

i build asterisk on pc with 3 network inerface eth0 (yyy.yyy.yyy.yyy) main public ip eth1 (xxx.xxx.xxx.xxx) seconf public ip used only for voip connection eth2 (zzz.zzz.zzz.zzz) local ip i config oh323 to bind eth1 interface i try make call from my local network -> Asterisk -> provider h323 wh

Re: [Asterisk-Users] fax - conversion problem

> "asterisk" == asterisk <[EMAIL PROTECTED]> writes: asterisk> The problem is in the tiff2ps, not in the ps2pdf. I asterisk> found that if I remove the -h and -w parameter asterisk> everything is OK My computer has a tiff2pdf command (from the libtiff-tools Debian package), so I

Re: [Asterisk-Users] SER and Asterisk

On 10/19/05, Yair Hakak <[EMAIL PROTECTED]> wrote: i do it this way because i want all the dialplan logic and CDR having to do with PSTN in asterisk, not SER. so, calls from the outside are adressed to [EMAIL PROTECTED]:5070 and hit asterisk. asterisk either sends them along to 5060, or handles

Re: [Asterisk-Users] TDMoE question

You can use MPLS which takes care all the point you had mentioned. appan kh - Original Message - From: "trixter aka Bret McDanel" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, October 19, 2005 9:54 AM Subject: [Asterisk-Users] TDMo

[Asterisk-Users] Call queuing question

Hi, Could I have clarification on the logic in app_queue which treats no answer as needing a retry? What I want to do is have all calls firstly always go to phone A, then if there is no answer make it call B or C in a round robin fashion. The obvious thing to do is put a penalty on B & C but then

[Asterisk-Users] what hw/OS to choose [please help]

Hi, First of all, I would like to say hello to everybody, it's my first post on the list. I'm building a pbx for a client and I need help/suggestions on what hardware and os to choose. I've read all I could find on the net, but still can't decide myself. Appart from signal switching, the main c

Re: [Asterisk-Users] sip rfc bye violated?

Matt Hess wrote: > Attached is a pcap of sip packets that pertain to another call similar > to the history shown.. it's hard to nail these down as it takes a lot of > time, patience and sifting through dumps. > Well, a pcap does not tell me how Asterisk reacts, sorry. That was what I wanted to see

Re: [Asterisk-Users] TDMoE question

On Wed, 2005-10-19 at 10:43 +0100, Appan KH wrote: > You can use MPLS which takes care all the point you had mentioned. > > appan kh Not entirely, at least not as I understand MPLS. MPLS will add a little bit of data which is used to route the traffic, it doesnt deal with encapsulating TDM data

Re: [Asterisk-Users] Menu/IVR and transfering to an extension after pressing an option number.

> > Here's what I'm trying to accomplish: > > Press 1 to transfer to extension > Press 2 for Directory > Press 0 for Operator > > Got directory and operator working. My problem is with transfering to an > extension after pressing 1. Asterisk keeps adding the 1 to the extension > that I need to

[Asterisk-Users] Url dialing

jsss> My suggestion would be the one-line eyeBeam phone under jsss> development. Check out support.xten.com. I checked a multiline versionof eyebeam: no url opening within the phone call, using this syntax: Dial(sip/399|||http://www.google.it) Could it be that only IAX2 supports this ? Tnx!

[Asterisk-Users] my SIPURA ATA does not make calls thru teliax

Greetings! Dear List, I had been making calls thru teliax for more than two month using my SIPURA ATA. but one day when my a/c balance fell around $ 18.00 (one month ago) and after that it could not make any calls thru teliax. until now I can not make calls. Mr. David said that their side is ok a

[Asterisk-Users] Asterisk on Slackware ...

Does anybody installed Asterisk on Slackware? It seems the installation went ok. But which config files do I have to look and edit first for the testing on two internal peers. Which free version of VoIP softphone is the best to use with asterisk?   Thanks ..   ___

Re: [Asterisk-Users] Asterisk on Slackware ...

Hello, We have 12 servers in production that run Asterisk on Slackware Linux they run beautifully. If you are starting out in Asterisk I suggest reading the recently completed Asterisk book(it's free in electronic format): http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 It's the

Re: [Asterisk-Users] more dids added to goiax.com

> > I made two attempts this morning to send some comments off list, and > both got returned due to some sort of spam filter, so would hope that > any future controls will not suffer from that inability to communicate. Maybe if you posted what the returned email said then he could remove or alte

[Asterisk-Users] IAX termination/DID provider in Panama?

Does anyone know of a IAX termination/DID provider in Panama? (507 country code). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asteri

Re: [Asterisk-Users] IAX only speech one way

Have you tried 'iax debug' and/or using ethereal to see what (if anything) either system might be spewing? > Thanks for your suggestion. > > Unfortunately, it didnt change anything, A can still not hear B, but B > can hear A, strange.. > > Michael > > 2005/10/19, R

Re: [Asterisk-Users] Priority jump in AEL

There is no way in AEL to specify the priority explicitly. To solve the problem use DB_EXISTS function. Here is an example from my dialplan:     if(${DB_EXISTS(Provider/${prov}/used)})     Set(MINUTES_USED=${DB_RESULT}); 

[Asterisk-Users] chan_capi.so: undefined symbol: ast_smoother_feed

Hello, i have installed Asterisk Version 1.2 from source on Sarge with a 2.6.8 Kernel. Then i did a "apt-get install libcapi20-2". When i start asterisk, i get this error: [ Booting...Oct 19 13:13:47 NOTICE[18363]: cdr.c:1160 do_reload: CDR simple

Re: [Asterisk-Users] Problems Calling PSTN PSTN FROM ASTERISK

> I terminated a call through SIP to a landphone i have the following > problems. > > 1.) asterisk gives a fake riming tone, it does not give the real tone from > the phone company. > > 2.) when I put the call on hold the on hold music is not very clear. > but when I talk the call quality is v

RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600

> Yupgot one running at home thanks to your WIKI. But for clients > moving forward, I need something a bit more mainstream. I'm > disappointed that the TE110P + adit 600 has been an issue on multiple > systems now, and that the software echo canceller has been a major > failure. > > It ma

[Asterisk-Users] Asterisk management portal

Does anybody have detailed instruction how to Install AMP? I have tried to install it using Installation Guide on their pages but I'm unable to satisfy AMP's PERL module dependencies. Thank you for your time. Tomislav ___ --Bandwidth and Colocation s

Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

On Tue, 18 Oct 2005, Matt wrote: > try sangoma card which has a very good echo cancel solution. Huh? I believe that the Sangoma uses the same zaptel echo cancellers that are used with the Digium cards. Steve ___ --Bandwidth and Colocation sponsor

Re: [Asterisk-Users] more dids added to goiax.com

On Wed, 2005-10-19 at 07:09 -0400, Steve Totaro wrote: > Or even use it? What is the point of having a DID that changes all the > time? > When I was younger this type of thing would have been just what my friends would have wanted. The ability to have a temporary disposable anonymous number for

[Asterisk-Users] Help with Dial Plan

Title: Help with Dial Plan Hi all. So far this list is proving it's worth, even on my first day using it! I hope that someone might know an easy solution to this one. I would like to create a dial plan which will allow me to have all extensions 6XXX cause a dial-out of my T1 interface (TE

Re: [Asterisk-Users] Call queuing question

Hello, if you use a mechanism like agents, * will know that there is nobody at the first level of penalty and route the call to the other level. A different approach could be to have a queue ring A for say 20 second, timeout, route the call to a second queue where B and C are. This should

[Asterisk-Users] Re: PRI echo issues: solvable?

Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: >On Tuesday 18 October 2005 12:18, Doug Meredith wrote: >> Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: >> >I've never seen that, it's always when we call out. Certain numbers will >> >always trigger it. 888-737-4787 (IPC Resistors, it dumps into an IVR

Re: [Asterisk-Users] Asterisk on Slackware ...

Dear Matt,   Thanks a lot for the asterisk book link. I am currently reading through the book. I'm also wondering that will it be able to use the FXO and FXS ports from Cisco 1760 router or is there some integrations to use with Cisco routers for Voice?   Regarding to the softphones, I will b

Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

On Wed, 2005-10-19 at 13:32 +0200, [EMAIL PROTECTED] wrote: > > On Tue, 18 Oct 2005, Matt wrote: > > > try sangoma card which has a very good echo cancel solution. > > > Huh? I believe that the Sangoma uses the same zaptel echo cancellers that > are used with the Digium cards. Unless you hav

Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

On Wed, 19 Oct 2005, Patrick wrote: > > Unless you have the Sangoma card with the hardware echo can on board. > So am I right in saying that the normal Sangoma uses the standard Zaptel software echo canceller - the same one that the Digium board uses? That's been my understanding, but people

Re: [Asterisk-Users] Call queuing question

Hi, Using agents would involve the user having to remember to login again every time they leave their desk as it would only be useful if they were auto-logged off ;) I've tried playing with the timeouts and have found that the timeout parameter to queue causes it to return to the dialplan after t

Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

> > Unless you have the Sangoma card with the hardware echo can on board. > > > > So am I right in saying that the normal Sangoma uses the standard Zaptel > software echo canceller - the same one that the Digium board uses? > > That's been my understanding, but people seem to keep popping up on

Re: [Asterisk-Users] Help with Dial Plan

On Wed, 19 Oct 2005, Dave Morrow wrote: > Hi all. So far this list is proving it's worth, even on my first day > using it! I hope that someone might know an easy solution to this one. > I would like to create a dial plan which will allow me to have all > extensions 6XXX cause a dial-out of my

Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

On Wed, 2005-10-19 at 14:06 +0200, [EMAIL PROTECTED] wrote: > > On Wed, 19 Oct 2005, Patrick wrote: > > > > > Unless you have the Sangoma card with the hardware echo can on board. > > > > So am I right in saying that the normal Sangoma uses the standard Zaptel > software echo canceller - the s

[Asterisk-Users] How can I signal a flash to PABX ...

Hi everybody, I have an Asterisk box connected locally to a PABX via analog and BRI extensions. Some remote VOIP phones are remotelly connected to this box, which acts as an IP gateway or better a remote PABX (analog extension 1 is connected to VOIP phone 1, extension 2 to VOIP 2, and so on.) The p

[Asterisk-Users] Fw: asterisk shutting down...

Hi,   Got the following messages log tonight... and Asterisk was down until I manually restarted it...   Any ideas?   Thank you Dov   Oct 19 03:40:18 WARNING[28005]: Avoided deadlock for 'SIP/raphael.pavanelli-f40b', 10 retries!Oct 19 03:40:28 NOTICE[28005]: Still have a call...Oct 19 03:40:5

[Asterisk-Users] SIP to IAX

Hello everybody, Is it possible to route "any" incoming SIP call (without authentication - register) from an Asterisk A to a remote Asterisk B(throught IAX2), transparently ? Otherwise said, I would like to pass any incoming SIP call from Asterisk A to Asterisk B without SIP need to be registered,

Re: [Asterisk-Users] SIP to IAX

YES - Original Message - From: "Frank Kostin" <[EMAIL PROTECTED]> To: Sent: Wednesday, October 19, 2005 8:58 AM Subject: [Asterisk-Users] SIP to IAX Hello everybody, Is it possible to route "any" incoming SIP call (without authentication - register) from an Asterisk A to a remote Aste

Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

[EMAIL PROTECTED] wrote: Tell me, are the Sangoma with hardware echo cancellation shipping? They are not yet shown on the website. The A104D begins shipping Monday and authorized Sangoma resellers are already accepting orders. See: http://shop.ifax.com/product_info.php?cPath=32_33&products

[Asterisk-Users] SIP CallerID

I am using a * w/a PRI for the TDM interface to telco. I am running Asterisk CVS-HEAD-05/29/05-03:59:44 All was working well until I needed a SIP ATA to be unlisted. in sip.conf, on the account I used: restrictcid=yes I am getting the callerID through though. I know that the ANI needs to be pa

RE: [Asterisk-Users] Asterisk Redundency

> Since I can't do that, what I've settled on is heartbeat + mon. > Heartbeat will monitor for a system level failure and switch to the backup machine if neccesary; and mon will watch the asterisk (or any > other) service and restart it and/or alert me if it fails. What kind of monitor are you

[Asterisk-Users] DNIS/DNID

Hi Is it possible with Asterisk to tell the called party which number was dialled by the caller?  Or in place of the number dialled have a description such as 'Sales' or 'Accounts'?  Ideally, I would like to show a description corresponding to the number dialled followed by CIDName.  How mi

[Asterisk-Users] Caller ID

If this question has an obvious answer forgive me, I'm a noob. I'm planning to make a system configured as below: POTS <---> FXO (400P) <--> Asterisk <---> FXS (400P) <> Analog phone The question I have is, if an incoming call from the POTS line has caller ID information, does/is/can that

Re: [Asterisk-Users] Caller ID

It just works has been my experience. Thanks, Steve - Original Message - From: "Michael J. Lynch" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, October 19, 2005 9:31 AM Subject: [Asterisk-Users] Caller ID > If this question has an o

Re: [Asterisk-Users] DNIS/DNID

exten => xx,2,SetCIDName(*-SALES-* ${CALLERIDNAME}) - Original Message - From: James Steven To: asterisk-users@lists.digium.com Sent: Wednesday, October 19, 2005 9:33 AM Subject: [Asterisk-Users] DNIS/DNID Hi Is it possible with Asterisk to tell the call

Re: [Asterisk-Users] Problem with select correct network interface (oh323)

it seems to me that your problem is not Asterisk configuration, but iproute configuration. Look in google about iproute and kernel routing tables. In order to help you, it would be desireable to know how are you dialing. Best Regards.On 10/19/05, Oleh Mukha <[EMAIL PROTECTED]> wrote: i build aster

Re: [Asterisk-Users] Caller ID

The question I have is, if an incoming call from the POTS line has caller ID information, does/is/can that information be passed onto the analog phone so it's caller id display will show the info? If so, is there anything I need to do to make this happen or does it *just work*? Thanks. It sh

Re: [Asterisk-Users] more dids added to goiax.com

Steve Totaro wrote: I made two attempts this morning to send some comments off list, and both got returned due to some sort of spam filter, so would hope that any future controls will not suffer from that inability to communicate. Maybe if you posted what the returned

Re: SV: [Asterisk-Users] Queues and call waiting indication

On Oct 18, 2005, at 9:08 AM, Adam Goryachev wrote: On Tue, 2005-10-18 at 14:35 +0200, [EMAIL PROTECTED] wrote: Hi, This issue has been discussed probably a million times on every asterisk forum in the world and I have the same problem too. Another problem you would have with the agents i

[Asterisk-Users] Connection question

Asterisk seems to be a very good peace of software, but i am interested to know if i can use plain ISDN cards with it, i mean use the isdn cards as a passthrough device between my alcatel pbx and voip users. thanks DLS - Projectos, Automaç

Re: [Asterisk-Users] Voicemail as an email attachement

Yes. I am interested. I will make provisions for the upload. How big are the files? Thanks BEN Goran Skular wrote: I changed my app_voicemail.c to work not with sendmail but with sendEmail that connects to any SMTP and sends email with attachment... It's dirty, but it works. If you are inte

Re: [Asterisk-Users] Recomendations for utility togenerateAsteriskconfiguration

On Oct 19, 2005, at 11:04 AM, asterisk wrote: AMP's dialplan and setup is quite complex. Requires, e.g, a number of AGIs.This is normally not the type of thing you'd like to hand-edit later after the initial adaptation to the target system.Who said an

Re: [Asterisk-Users] Polycom IP501 and record on demand

Matt Gibson wrote: You could also take a look at features.conf, and use ** for blind transfers, ## for attended transfers, *0 for recording, and *1 to hangup. I haven't tried mapping them to polycom buttons, but there was recently a discussion about that, just this week you can search the arc

RE: [Asterisk-Users] DNIS/DNID

That worked great. Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve TotaroSent: 19 October 2005 14:45To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] DNIS/DNID exten => xx,2,SetCIDName(*-SALES-* ${CALLERIDNAME}) -

RE: [Asterisk-Users] Polycom IP501 and record on demand

I probably can't provide any better information for you, however, have you looked through the Polycom configuration files. The button mappings are there. I haven't spent much time with it so I can not attest to what you can map them to do. Hope this helps you a little. -Jonathan -Original

[Asterisk-Users] Trunk Dialing rules

Hi i have posted before about this problem, and have had several suggestions, that i can use contexts to overcome this. The situation. [EMAIL PROTECTED] 1.5 I have 3 sets of users say sales, admin and tech with the numbers sales200 201 admin 202 203 tech 204 205 They all need to be

[Asterisk-Users] E1 PRI error: "!! Got I-frame while link state 2" and "!! Got a UA, but i'm in state 1" (long)

Original Message Subject: E1 PRI error: "!! Got I-frame while link state 2" and "!! Got a UA, but i'm in state 1" Date: Wed, 19 Oct 2005 23:46:01 +0800 From: Dinesh Nair <[EMAIL PROTECTED]> To: Asterisk Users Mailing List - Non-Commercial Discussion , Asterisk on BSD discuss

Fwd: Re: [Asterisk-Users] IAX only speech one way

IAX may be as bad as what we are doing?Note: forwarded message attached.--- Begin Message --- Mir wrote: > Thanks for your suggestion. > > Unfortunately, it didnt change anything, A can still not hear B, but B > can hear A, strange.. > I had the same problem with one of my IAX providers in A

[Asterisk-Users] Caller-ID via database lookup

Hey everybody, I'm having issues with one of our facilities, concerning caller-id. The system is a Definity that hits a second Definity. The 2nd Definity trunks the call to my Asterisk server via a TE110P. I can only get Caller-ID name. Nothing in the From: field. I thought I would be ab

Re: [Asterisk-Users] Voicemail as an email attachement

On Tue, Oct 18, 2005 at 11:22:52PM -0500, Ben Brown wrote: > I have configured the voicemail.conf file as per the wiki to email > voicemails as an attachment. I cannot find any instructions/locations to > set the outgoing server login information. Furthermore, I can get no > emails from asterisk

[Asterisk-Users] Re: Polycom IP501 and record on demand

Hi Matthew - You could also take a look at features.conf, and use ** for blind transfers, ## for attended transfers, *0 for recording, and *1 to hangup. I haven't tried mapping them to polycom buttons, but there was recently a discussion about that, just this week you can search the archives

RE: [Asterisk-Users] Caller-ID via database lookup

I have my Definity attached to my Asterisk box with a PRI Trunk. The guides and seemingly most people say to use a tie type connection, however I did not get correct caller-id and setup until I: 1) Set the trunk-group on the Definity to isdn 2) Carrier/Medium to PRI 3) Trunk group numbering forma

Re: [Asterisk-Users] Re: Polycom IP501 and record on demand

That is perfect for one-button remaps! I guess I migrated away from one-button features in * but I see the light now. The trouble Matthew and I were having was to stimulate presses of more than one button in a sequence -- "SpeedDial" function was the only one I could find that was close, but t

Re: [Asterisk-Users] Caller-ID via database lookup

O'Connor, Jonathan wrote: I have my Definity attached to my Asterisk box with a PRI Trunk. The guides and seemingly most people say to use a tie type connection, however I did not get correct caller-id and setup until I: 1) Set the trunk-group on the Definity to isdn 2) Carrier/Medium to PRI

Re: [Asterisk-Users] DID setup from goiax.com

Trixter: Thanks for the guide to setting this up:... I have tried the below configuration with my settings, and when I place /goiax-in after my register command, my register statement fails. If i remove it. I get a Rejected connect attempt from goiax's server IP, trying to reach 's@' I have put

[Asterisk-Users] uable to establish link between asterisk to external phone

  Hi,   I am new Asterisk. I configured asterisk1.5 and be able to communicate from iaxComm dial pad to external computer i.e out side my router/LAN. When I make call from iamComm of external computer to my cell phone, I am getting the ring but not able to listen voice on both sides. Do I need to m

Re: [Asterisk-Users] Agent recording and muxmon

Julian Lyndon-Smith wrote: Torrow your time I presume - it's today in the uk:). Will this be in 1.2, or is it a post 1.2 ? It will be in 1.2. I don't understand why they would be incompatible changes - could you not add a MuxMon facility as another option. e.g. in agents.conf: RecordAgentC

[Asterisk-Users] NEWBIE HELP : chan_zap.c: Exception on 16, channel 1, call not being picked up on incoming X1-100P zap

I am running [EMAIL PROTECTED] ( asterisk 1.2beta1 with two X100P cards ) on centos 4.1 box with a 2.6.12 kernel. I ran genzaptelconf and added two trunks for each of the devices however the incoming calls when I ring just get ignored. asterisk -r tells me that it just gets hangupcall, and

Re: [Asterisk-Users] Asterisk management portal

Tomislav Parčina wrote: Does anybody have detailed instruction how to Install AMP? I have tried to install it using Installation Guide on their pages but I'm unable to satisfy AMP's PERL module dependencies. Please post to the amportal-users list: http://lists.sourceforge.net/lists/listinfo/

Re: [Asterisk-Users] DID setup from goiax.com

for the incoming context put your goiax.com phone number not the free DID number but the other one.On 10/19/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: Trixter:Thanks for the guide to setting this up:... I have tried the belowconfiguration with my settings, and when I place /goiax-in after my

Re: [Asterisk-Users] DID setup from goiax.com

That is What I stated in the email.. my GOIAX #. not the DID #. That is not the issue. > for the incoming context put your goiax.com phone > number > not the free DID number but the other one. > > On 10/19/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: >> >> Trixter: >> >> Th

Re: [Asterisk-Users] DID setup from goiax.com

Replace [goiax] with [PHONENUMBER] username= don't work for users in IAX channel. On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote: > That is What I stated in the email.. my GOIAX #. not the DID #. > > That is not the issue. > > > for the incoming context put your goiax.com

[Asterisk-Users] Re: Polycom IP501 and record on demand

Hi Mojo - The trouble Matthew and I were having was to stimulate presses of more than one button in a sequence -- "SpeedDial" function was the only one I could find that was close, but this opens a new call appearance for the call rather than just playing the dtmf over the open one. Yeah,

Re: [Asterisk-Users] DID setup from goiax.com

That did it... Thank you. putting / after the register line caused it to not register any more... and i would get error server1.goiax.com/ could not be found. anyways.. thanks for your help guys :) > Replace > [goiax] > with > [PHONENUMBER] > > username= don't work for users in IAX channel

[Asterisk-Users] unable to make connectivity between asterisk to external phone

Hi All,   I am new to Asterisk. I configured asterisk and be able to communicate from iaxComm dial pad to external computer i.e out side my router/LAN. When I make call from iamComm of external computer to my cell phone, I am getting the ring but not able to listen voice . Do I need to make any sp

[Asterisk-Users] teliax audio issues - response

For those on the list using iax with teliax.com, the intermitant one-way audio problem that I reported to them received the following response: "We currently use our own version of 1.2 with our own patches on our boxes and the iax code is updated. You cannot use jitterbuffers with g729 or gsm as

Re: [Asterisk-Users] initiate call recording from phone.

Well... I don't know anything about [EMAIL PROTECTED] I know even more nothing about dialparties.agi... but I can summarize http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial for you: Let's say you want to call out on a PSTN line. A command such as the following will be in your o

Re: [Asterisk-Users] sip rfc bye violated?

Matt Hess wrote: > I should have mentioned that I can't do a full sip log.. with several > calls a second whipping through this system it's almost impossible to > weed out the info for the proper call.. and usually I don't see the dead > channel until well after the fact. > Looked at this with coo

[Asterisk-Users] possible bug, what do you think?

We recently changed file formats on our server to wav49 from gsm. Several users had saved messages in gsm format. When a user attempts to forward an old message to a user and they prepend the message with a recording, the process seems to be flawed. From what I can tell, the prepend message is re

Re: [Asterisk-Users] sip rfc bye violated?

Matt Hess wrote: > I should have mentioned that I can't do a full sip log.. with several > calls a second whipping through this system it's almost impossible to > weed out the info for the proper call.. and usually I don't see the dead > channel until well after the fact. http://bugs.digium.com/vi

RE: [Asterisk-Users] Help with Dial Plan

Thanks Steve. It almost works, but never dials the extension. Also, is there a way I could mute the line while the remote attendant comes on? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 <

Re: [Asterisk-Users] uable to establish link between asterisk to external phone

kotesh m wrote: > > Hi, > > I am new Asterisk. I configured asterisk1.5 and be able to communicate That is amazing. You are new and already at version 1.5. I have been around for a while and only reached 1.1dev, working on 1.2 :-) Guess I have some catching up to do... /O ;-) (Sorry, could no

Re: [Asterisk-Users] DID setup from goiax.com

I dont know then that was cut and paste from what I have working ... maybe actual log dumps of the error? On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote: > That is What I stated in the email.. my GOIAX #. not the DID #. > > That is not the issue. -- Trixter http://www.0xdecafbad.co

Re: [Asterisk-Users] possible bug, what do you think?

Andy Goss wrote: error. The end result is that the user gets a new message envelope in their INBOX (msg.txt) but there is no associated .WAV file to go along with it. The desired behavior here is to a) notify the user who is attempting to forward this message that the process failed so tha

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