Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

Hello Phil, On Saturday, April 23, 2016, 11:11:29 PM, you wrote: > Actually, this is now sorted. It turns out the latest recommended > configs on the A wiki had peer vs. user confusion. On correcting > this, all was well. I'm glad you found it. It look me a while to track down that problem when

Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

On Sat, 23 Apr 2016 22:45:32 +0100 Julian Beach wrote: > Hello Phil, > > I have a couple of lines with A, and I have not been having any > problems recently. When I have had similar problems in the past, it > has been an issue with the SIP config. I originally had a number

Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

Hello Phil, On Saturday, April 23, 2016, 12:19:15 PM, you wrote: > I have checked that the username and password in my config agree both > ends, and have even tried changing them. > The bulk of my calls come in on A, so I am obviously trying to find > out what has gone wrong. No-one else is

[asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

I have service with both VoIPtalk.org and Andrews & Arnold (aa.net.uk). VoIPtalk calls are unauthenticated and reach me fine, but Andrews & Arnold calls are authenticated. The last call I successfully received was on Tuesday afternoon. Initially, A were for some odd reason not sending calls to my

Re: [asterisk-users] Incoming calls get 488 error

On 8/21/15 6:45 PM, Technical Support wrote: I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring at the handset. I captured the SIP traffic and see that my M65 is replying with an 488 not acceptable here. From what I read this is usually codec related but

[asterisk-users] Incoming calls get 488 error

I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring at the handset. I captured the SIP traffic and see that my M65 is replying with an 488 not acceptable here. From what I read this is usually codec related but both asterisk and the M65 are set for ulaw as

Re: [asterisk-users] Incoming calls get 488 error

- From: Technical Support [supp...@telium.ca] Received: sexta-feira, 21 ago 2015, 19:46 To: asterisk-users@lists.digium.com [asterisk-users@lists.digium.com] Subject: [asterisk-users] Incoming calls get 488 error I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring

[asterisk-users] Incoming calls to a GSM gateway SIP/2.0 401 Unauthorized response when dial 7777 to Asterisk

Hello: I'm newbie in asterisk, please help me. My context is as follows: 192.168.4.2 -- Asterisk 11.13.1 complied from source 192.168.4.4 -- Yeastar NeoGate TG100 GSM gateway When I call from a GSM cell phone, my TG100 GSM gateway answers and dials extension (configured as a hotline on

Re: [asterisk-users] incoming calls fall into echo test mode

On Saturday 19 Jul 2014, Norman Molhant wrote: I tried many things on our FreePBX box and found out the problem seems somehow linked with the customer's extension (or phone number), not his inbound route (changing the latter has no effect on the problem). Creating a new extension with

[asterisk-users] incoming calls fall into echo test mode

Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each with its own phone number, with no problem at all, except for one (and only one) subscriber who has this problem: his outgoing calls are ok, but when someone dials his phone number (be it from our network or

Re: [asterisk-users] incoming calls fall into echo test mode

is being misrouted in the dialplan From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Norman Molhant ad...@csur.ca Sent: Saturday, July 19, 2014 10:43 AM To: Asterisk Users List Subject: [asterisk-users

Re: [asterisk-users] incoming calls fall into echo test mode

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant Sent: Saturday, July 19, 2014 10:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming calls fall into echo test mode Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each

Re: [asterisk-users] incoming calls fall into echo test mode

To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming calls fall into echo test mode Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each with its own phone number, with no problem at all, except for one (and only one) subscriber who has

Re: [asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

Just out of curiosity, what country are you in? I agree with the others in this thread, this seems very bizzare that the telco requires you to do SS7 for dialup connections. I would ask them for specifics about the legal issues with what you are doing - it sounds to me like they are just trying

Re: [asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

Matt, We are located on Costa Rica and so far there's just 1 TELCO running the industrym with the CAFTA treatment the carrier had to open for interconnection but they get to define the ground rules for the interconnection. They are arguing ISDN is and end customer circuit and you cannot use it

[asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted

Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??

To: asterisk-users@lists.digium.com Subject: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible?? Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem

Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??

:* [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible?? Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out

Re: [asterisk-users] Incoming calls through SS7 for datamodemtransmissions - possible??

. Cary -- *From:* José Pablo Méndez Soto [mailto:aux...@gmail.com] *Sent:* Wednesday, November 24, 2010 8:34 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* ca...@usawide.net *Subject:* Re: [asterisk-users] Incoming calls through SS7

[asterisk-users] Incoming calls

Hi all, I'd like that each analog trunk of my TDM410p was received in different extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a different context and in my extensions.conf, under [default] I put such contexts and an especific estension to answer it. therefore,

Re: [asterisk-users] Incoming calls

On Thu, 18 Nov 2010, Flavio Miranda wrote: I'd like that each analog trunk of my TDM410p was received in different extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a different context and in my extensions.conf, under [default] I put such contexts and an

Re: [asterisk-users] Incoming calls

MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Thu, 18 Nov 2010 11:53:26 -0800 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Incoming calls On Thu, 18 Nov 2010, Flavio Miranda wrote: I'd like that each analog trunk of my TDM410p

Re: [asterisk-users] Incoming calls

On Thu, 18 Nov 2010, Flavio Miranda wrote: Looking to dahdi show channles , I realized  that all the trunks was in the same context. So, I have changed  this and everything works! That's why I prefer to work from what Asterisk parsed the file as, not what the poster thinks :) -- Thanks in

[asterisk-users] Incoming calls

Hi all, After a lot of trouble with a TE110p working with mfcr2 , brazil variant, everything looks great,but I can not go out of my calls. When I try I receive the following log: == Using SIP RTP CoS mark 5-- Executing [33220...@local:1] Dial(SIP/4804-001a, DAHDI/g11/33220567,,T)

[asterisk-users] Incoming calls coming into default context

My SIP-provider sends my a SIP-invite like this : INVITE sip:329298y...@80.xx.xx.69:5060 SIP/2.0 Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c Max-Forwards: 70 From: sip:321445x...@80.xx.xx.69;tag=f395877e02bf8eb2fd8f5a0e To: sip:329298y...@80.xx.xx.69 Call-ID:

Re: [asterisk-users] Incoming calls coming into default context

21 dec 2009 kl. 12.00 skrev jonas kellens: My SIP-provider sends my a SIP-invite like this : INVITE sip:329298y...@80.xx.xx.69:5060 SIP/2.0 Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c Max-Forwards: 70 From:

[asterisk-users] Incoming Calls via SIP Trunks

Hello, i have one question regarding incoming SIP INVITES. I have a testbed where i have 5 extnsions : 6001 - 6005 Domain : domainA.com Then i have configured a sip trunk, where my PBX registers to a foreign SIP Proxy. All is working fine, until following scenario: Incoming call from [EMAIL

Re: [asterisk-users] Incoming calls on zaptel not answered.

On July 14, 2008 08:24:33 pm Jose Flores Galicia wrote: After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri, zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop working. THis isn;t going to fix your problem... but just FYI, you don't need to install

Re: [asterisk-users] Incoming calls on zaptel not answered.

Hi, I need libpri, because I have a TE110P E1 with a PRI ISDN service. 2008/7/15 Matt Watson [EMAIL PROTECTED]: On July 14, 2008 08:24:33 pm Jose Flores Galicia wrote: After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri, zaptel, the incoming calls to a TDM400P REV I,

Re: [asterisk-users] Incoming calls on zaptel not answered.

Thank you, yes, I changed the PCI Slot and it's the same, I get a used card from a customer with 2 FXO, same REV, that board was working on the customer server, put it on mine, and stop working. I put my board on his server and the board is working perfectly. I had not test outgoing calls on

[asterisk-users] Incoming calls on zaptel not answered.

After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri, zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop working. The board is working, I tested in another server with the 1.2.13 asterisk version. When a call is incoming, I do a ztmonitor to check the rx

Re: [asterisk-users] Incoming calls on zaptel not answered.

Hi Jose - After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri, zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop working. The board is working, I tested in another server with the 1.2.13 asterisk version. Also changed the pci slot where the board

Re: [asterisk-users] incoming calls through callcentric sip account!!

Le vendredi 20 juin 2008, RoLaNd RoLaNd a écrit : Hi all, i've recently acquired a callcentric account. i've perfectly setup my sip.conf and extensions.conf to make outgoing calls. Well, I had the same problem and had to debug. In fact for some reason, and it's a bit hackward, incoming

[asterisk-users] incoming calls through callcentric sip account!!

Hi all, i've recently acquired a callcentric account. i've perfectly setup my sip.conf and extensions.conf to make outgoing calls. but the problem is with incoming calls! when i call in, asterisk doesnt even see the incoming call! how is tht possible! please see the following my config:

Re: [asterisk-users] Incoming calls not being answered by asterisk

From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Incoming calls not being answered by asterisk The first thing to do is type sip debug on the console and place the call from the Sipura. If you get a bunch of SIP messages flashing down your console you

[asterisk-users] Incoming calls not being answered by asterisk

Hello all, ive got the following setup currently: __Sipura 3102-PSTN | Lan | | |__asterisk i configured both asterisk and pstn to be able to receive/make calls through each other using sip of course.. but the thing is i want asterisk that when it receives an

Re: [asterisk-users] Incoming calls not being answered by asterisk

The first thing to do is type sip debug on the console and place the call from the Sipura. If you get a bunch of SIP messages flashing down your console you know the call is reaching Asterisk and it's most likely going to be an issue authenticating the call or a problem in your dial plan. If no

Re: [asterisk-users] Incoming calls not being answered by asterisk

Ciao Roand I think you should buy a book and do some reading to build up your knowledge. but in the meantime try something like this in the dialplan (extensions.conf) exten = PSTN,1,Answer() ; Answer inbound calls or internal miss-dials exten = PSTN,2,Playback(silence/1) exten =

[asterisk-users] Incoming calls on PSTN trunk not disconnected (bsnl, india)

I am trying to configure Asterisk for BSNL, india network. I have successfully configured it for outgoing calls. When any outside number make any call to trunk then it receives the call properly but when the call is disconnected by inside extension then outside phone does not get a busy tone.

[asterisk-users] Incoming Calls

I am having a problem that I would like to verify if someone could help...I am using bandwith.com as my SIP TRUNK provider. When I place the phone number in the DID number field ( using Elastix) it gives me an error message stating the phone number I dialed is not in service. When I leave the DID

Re: [asterisk-users] Incoming Calls

Paulo Pinheiro wrote: I am having a problem that I would like to verify if someone could help…I am using bandwith.com as my SIP TRUNK provider. When I place the phone number in the DID number field ( using Elastix) it gives me an error message stating the phone number I dialed is not in

Re: [asterisk-users] Incoming Calls

Hi Mr. Paulo, Could you please explain this situation in a more detailed way to see how can we help you? Regards, Paulo Pinheiro wrote: I am having a problem that I would like to verify if someone could help...I am using bandwith.com as my SIP TRUNK provider. When I place the phone

Re: [asterisk-users] Incoming Calls

AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Incoming Calls I am having a problem that I would like to verify if someone could help.I am using bandwith.com as my SIP TRUNK provider. When I place the phone number in the DID number field ( using Elastix) it gives me an error

Re: [asterisk-users] Incoming Calls

Hi Paulo, Make sure your DID number is in the e.164 format, ie, +15551234567. I had the same issue with bandwidth.com and that fixed the problem. HTH, Zaheer Zaheer is right. Everything from bandwidth is 164 format. So you need the +15551234567 in the dial plan as well as in your

Re: [asterisk-users] Incoming Calls

PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jose P. Espinal Sent: Wednesday, January 02, 2008 11:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming Calls Hi Mr. Paulo, Could you please explain this situation in a more detailed way

Re: [asterisk-users] Incoming Calls

_ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Pinheiro Sent: Wednesday, January 02, 2008 10:21 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Incoming Calls I am having a problem that I would like to verify if someone could help...I am using

Re: [asterisk-users] Incoming Calls

used. Jonn _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Pinheiro Sent: Wednesday, January 02, 2008 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming Calls Hi Jose, I apologize for the lack

Re: [asterisk-users] Incoming Calls

in the User Details section of the trunk set up? Thanks much, Paulo From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonn R Taylor Sent: Wednesday, January 02, 2008 12:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming Calls

Re: [asterisk-users] Incoming Calls

Hi John, I have copied your changes in the Peer Details section of the trunk set up…then I went ahead and added the DID number in the Income Routes but still did not work. I tried the number alone and also tried adding the + sign in front of it. Do you think we should have any changes in the

[asterisk-users] Incoming calls

Hello I have a question about incoming calls on TDM400P cards. I want to know why an incoming call appear in a sorpresive way on a phone that I pickup to call out. I am using ChanIsAvailable to check those lines ( Zap channels )that are free. I have four lines connected to my TDM400P card and when

Re: [asterisk-users] Incoming calls

Glare that's what it's called, if the number you advertise as your business number is zap/1 then use zap/G1 to dial out, otherwise use zap/g1 to dial out. This will reduce but not eliminate the problem. http://www.telos-systems.com/techtalk/gldefs.htm#Glare On 10/18/07, Gustavo Gonzalez [EMAIL

Re: [asterisk-users] incoming calls in SIP

-users@lists.digium.com Sent: Saturday, August 18, 2007 3:25 PM Subject: [asterisk-users] incoming calls in SIP Hi, when I try to call in it tells me: NOTICE[11664]: chan_sip.c:10637 handle_request_invite: Failed to authenticate user 585415198 sip:[EMAIL PROTECTED];tag=as18abefe8 Can

[asterisk-users] incoming calls in SIP

Hi, when I try to call in it tells me: NOTICE[11664]: chan_sip.c:10637 handle_request_invite: Failed to authenticate user 585415198 sip:[EMAIL PROTECTED] sip:[EMAIL PROTECTED];tag=as18abefe8 Can someone help me out of this? I have Asterisk 1.2 on the Ubuntu 7.04. Outcoming and internal calls

[asterisk-users] Incoming calls don't arrive for correct number

I have an asterisk box registering 100 numbers on a voip provider. Numers are: 2546.1000 to 2546.1099 My problem is that every incoming call arrived to number 2546.1099 that is the last number to register on voip provider. The correct is call arrive in destination number. See this exaple: I call

Re: [asterisk-users] Incoming calls don't arrive for correct number

your problem is that you need to handle this in your dialpan to achieve which DID has been dialed! look for SIPGETHEADER application on asterisk, you shoul look for variable to where it comes the DID On 11/27/06, Frederico Madeira [EMAIL PROTECTED] wrote: I have an asterisk box registering 100

[asterisk-users] Incoming calls, identify

Just delving into asterisk, using trixbox 1.2 and a TDM400p. The card will have two FXO and two FXS modules. Two incoming analog lines, which need to be treated as distinct entities. Meaning, for example, line 1= company1, line2=company2, or line 1= home line, line2=business line. In my

Re: [asterisk-users] Incoming calls, identify

On Wed, Sep 20, 2006 at 08:48:17AM -0400, joea, j4computers wrote: Just delving into asterisk, using trixbox 1.2 and a TDM400p. The card will have two FXO and two FXS modules. Two incoming analog lines, which need to be treated as distinct entities. Meaning, for example, line 1= company1,

Re: [asterisk-users] Incoming calls, identify

Jay R. Ashworth[EMAIL PROTECTED] Wrote on: 9/20/2006 4:00 PM: On Wed, Sep 20, 2006 at 08:48:17AM -0400, joea, j4computers wrote: Just delving into asterisk, using trixbox 1.2 and a TDM400p. The card will have two FXO and two FXS modules. Two incoming analog lines, which need to be treated as

RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p

Discussion Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110pPaul C wrote: I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels

[Asterisk-Users] Incoming calls

Hi, I run an asterisk server. The configuration is very basic. Here is my question : When someone calls my phone line, which is connected to an FXO card, asterisk is answering using the context : ; Incoming calls goes to this default context : [incoming-rtc] include = postes-sip ; exten =

Re: [Asterisk-Users] Incoming calls

When a friend calls, I would like for him to enter a 4 digit password in order to access to a sub-menu, if no password is entered, then the welcome msg is said ... Any hints on how to do that ?? In your incoming-rtc context, define an extension (let's say 1234) exten =

Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p

[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 01, 2006 5:15 PM Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p Paul C wrote: I am running Asterisk 1.0.9 and have been running all my calls

Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p

Discussion' asterisk-users@lists.digium.com Sent: Sunday, March 05, 2006 6:52 AM Subject: RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of the site. It is sending CRC errors )to Telsta, drops all calls once

RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p

Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul C Sent: Wednesday, 1 March 2006 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p Paul C wrote: I am running Asterisk 1.0.9 and have

Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p

- Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Sunday, March 05, 2006 7:52 AM Subject: RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of the site. It is sending CRC errors )to Telsta

[Asterisk-Users] incoming calls dropout on PRI over TE110p

I am running Asterisk 1.0.9 and have been running all my calls througha VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few

Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p

Paul C wrote: I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped

Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p

Paul C wrote: I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are

[Asterisk-Users] Incoming Calls Getting Crossed - Weird

Hey, I got a weird one for you guys,I am running vanilla 1.2.4 and have all incoming calls come in as SIP from teliax. Twice over the past week 2 callers who have called in around the same time end up talking to each other instead of going through the ivr or at some point during the IVR.

RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird

You have stumbled across the new undocumented feature app_MakeAFriend or posably app_MakeAConsultantCrazy. Look to see if you have any SNOM phones, Hey, I got a weird one for you guys,I am running vanilla 1.2.4 and have all incoming calls come in as SIP from teliax. Twice over

RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird

PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander LopezSent: Monday, February 20, 2006 6:14 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird You have stumbled across the new undocumented feature app_MakeAFriend

RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird

nt: Monday, February 20, 2006 6:38 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird LMAO! app_PatientDatingService Yes I have all Snom 360's, are you thinking the problem isnt asterisk but instead

RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird

DiscussionSubject: RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird It is not the firmware but a setting. "Call Join on Xfer (2 calls)" Make sure that is is set to OFF. SNOMS are great ophone but 'features' like this drive me crazy. Alex From: [EMAIL

[Asterisk-Users] Incoming calls grind to a halt

Hi there everybody, We are running Asterisk 1.2.1 with a TE410P card attached to one PRI ISDN line, and many SIP phones. Yesterday we ended up in a situation where all incoming calls were giving the engaged tone. Every time some tried to ring in we got: Jan 4 14:56:32 WARNING[896]

Re: [Asterisk-Users] Incoming calls via CAPI and AVM Fritz Card

Hi, I'm not really sure if this helps you, but as far as I remember, the diastring with chan_capi-cm-0.6 is not CAPI/g1/0299546476:b${EXTEN},30,r but CAPI/ggroup/destination[/params] or in your case CAPI/g1/${EXTEN}/b,30,r. To set your CallerPresentation, use the SetCallerPres() in your

Re: [Asterisk-Users] Incoming calls via CAPI and AVM Fritz Card

On Wed, 26 Oct 2005, Esteban Guana-Jarrin wrote: Can anyone please provide some help. I have installed an AVM fritz card on an asterisk box ([EMAIL PROTECTED] version 1.5). I have installed the card driver and chan_capi-cm-0.6. According to the installations guide I can now see that the CAPI

Re: [Asterisk-Users] Incoming calls via CAPI and AVM Fritz Card

anything except a tone dropping the call. Armin, I will appreciate if you can put me in the right direction? Cheers PolAus From: Armin Schindler [EMAIL PROTECTED] To: Esteban Guana-Jarrin [EMAIL PROTECTED] CC: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Incoming calls via CAPI

[Asterisk-Users] Incoming calls via CAPI and AVM Fritz Card

Can anyone please provide some help. I have installed an AVM fritz card on an asterisk box ([EMAIL PROTECTED] version 1.5). I have installed the card driver and chan_capi-cm-0.6. According to the installations guide I can now see that the CAPI channel in asterisk is up, *CLI capi info Contr1:

[Asterisk-Users] Incoming Calls causing Protocol Error (6)

Hi Everyone, Got a setup as follows: Telco Siemens HiCom 300E Asterisk1 IAX2 Trunk Asterisk2 Siemens HiPath 4xxx The solution works except for one problem. Incoming calls from the telco get redirected to the Asterisk1 box with the correct extention, only if there is a

[Asterisk-Users] Incoming calls

Hi, stupid question: how can I let to call an extensions from outside? Untill now, I've just the possibility to call our number and then, after the system answer, dial the extension. My sistem is like this: SER - internal extensions Asterisk - incoming/outgoing gateway. FaberK -- .:FaberK:.

[Asterisk-Users] incoming calls

hi ; our * handle good the outgoing calls but 4 incaming calls we have this msg: Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)...someone have an idea ??, thx in advance, CaraMail met en oeuvre un nouveau Concept de Sécurité Globale à partir de 1,49

[Asterisk-Users] incoming calls

hi ; our * handle good the outgoing calls but 4 incaming calls we have this msg: Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)...someone have an idea ??, thx in advance, CaraMail met en oeuvre un nouveau Concept de Sécurité Globale à partir de

Re: [Asterisk-Users] incoming calls

youa re using -v option multiple times at startup. That message is perfectly fine. ali kia wrote: hi ; our * handle good the outgoing calls but 4 incaming calls we have this msg : Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... someone

Re: Re: [Asterisk-Users] incoming calls

Abrego [EMAIL PROTECTED] A: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Objet: Re: [Asterisk-Users] incoming calls Date: Fri, 22 Jul 2005 06:53:19 + youa re using -v option multiple times at startup. That message is perfectly fine. ali

Re: Re: [Asterisk-Users] incoming calls

On Fri, Jul 22, 2005 at 04:41:01PM +, salahssaid2.salah wrote: From: Andres Tello Abrego [EMAIL PROTECTED] Date: Fri, 22 Jul 2005 06:53:19 + youa re using -v option multiple times at startup. That message is perfectly fine. And thus see quite a few messages that are not

[Asterisk-Users] Incoming calls from BudgetPhone.nl

(this time with subject) Hello, I’m trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip

Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl

On 19:16, Sun 10 Jul 05, Peter Raaijmaker wrote: (this time with subject) Hello, I?m trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my

RE: [Asterisk-Users] Incoming calls from BudgetPhone.nl

- Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl Long short, Maybe X-Ten has an stun relay setup and Asterisk doesn't? Rene Kluwen Chimit (this time with subject) Hello, I’m trying to get Asterisk to accept incoming calls from budgetphone.nl

Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl

Try using insecure=very in your peer definition. That makes asterisk not require authentication from your peer if comes from the ip address give in host=xxx.xxx.xxx.xx directive. That helped me receiving calls from my sip provider, which had exactly the same problem. Julian. On 7/10/05, Peter

RE: [Asterisk-Users] Incoming calls from BudgetPhone.nl

PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl Try using insecure=very in your peer definition. That makes asterisk not require authentication from your peer if comes from the ip address give in host

Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl

Same here, Audio quality is ok. SIP registration sucks. The helpdesk makes me believe that it is a problem on my side: Your asterisk doesn't respond to a sip request in time. But I have no problems with any other provider, except with Budgetphone. I am not even getting a SIP request, so how do I

Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl

On 20:42, Sun 10 Jul 05, Rene Kluwen wrote: Same here, Audio quality is ok. SIP registration sucks. The helpdesk makes me believe that it is a problem on my side: Your asterisk doesn't respond to a sip request in time. But I have no problems with any other provider, except with Budgetphone.

[Asterisk-Users] Incoming Calls

I have 2 4-port Digium FXS cards in my system. I would like to play a different recording based on which trunk rings. Any pointers? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Incoming Calls

On Mon, 2005-06-06 at 15:25 -0400, David Sampson wrote: I have 2 4-port Digium FXS cards in my system. I would like to play a different recording based on which trunk rings. Any pointers? Thanks This is really a no brainer if you read the documentation. Simple have each

[Asterisk-Users] Incoming calls picked-up then simply hanged-up

I don't think my first posting went thru. I am trying to set up Asterisk for the first time. I am new to this. I am using [EMAIL PROTECTED] I have a TDM400P with one FXO and one FXS The system is working for outgoing calls and if I test incoming calls using . But when doing an actual call

[Asterisk-Users] Incoming calls picked-up then simply hanged-up

You should put your asterisk into verbose mode using asterisk -c or if you are using a server asterisk -r and you can trace out what happens and it will be in the log file called full in the /var/log/asterisk directory and then you can probably figure out what happened. Your incoming call

RE: [Asterisk-Users] Incoming calls picked-up then simply hanged-up

Thanks I will give that a try. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: Thursday, May 12, 2005 9:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Incoming calls picked-up then simply

RE: [Asterisk-Users] Incoming calls picked-up then simply hanged-up

PROTECTED] On Behalf Of John covici Sent: Thursday, May 12, 2005 9:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Incoming calls picked-up then simply hanged-up You should put your asterisk into verbose mode using asterisk -c or if you are using

Re: [Asterisk-Users] Incoming calls picked-up then simply hanged-up

Are you sure you have context=from-pstn in your zapata.conf for the fxo channels? Julian. On 5/12/05, fhunter [EMAIL PROTECTED] wrote: I don't think my first posting went thru. I am trying to set up Asterisk for the first time. I am new to this. I am using [EMAIL PROTECTED] I have a

RE: [Asterisk-Users] Incoming calls picked-up then simply hanged-up

Yep. Check context and it point to from-pstn Any other ideas. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M. Sent: Thursday, May 12, 2005 11:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users

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