[asterisk-users] zaptel debugging
Hi, How to enable zaptel debugging? I need to see reverse polarity messages. Thank you, Imran -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Server Configuration for E1's
Dear Users, I am fairly new to Digium and Asterisk. I wanted to know that if I use the Digium product THREE Digium Wildcard TE412Ps (Quad E1 Card) how many calls can I handle simultaneously. I want to use the cards with the following Configurations: Intel® Xeon 3.00GHz/800MHz, 2M Processor 1GB PC2-3200 DDR2 SDRAM (2x512MB + 2X1GB) memory Integrated Dual Channel Ultra320 SCSI Adapter NC7781 Single Port PCI-X embedded NIC Hot plug drive cage - Ultra3 (6X1) High Speed IDE CD-ROM Drive 72.8GB Pluggable Ultra320 SCSI 15,000 rpm (1) Universal Hard Drive Asterisk Business Edition 3 X TE412P I have a requirement of handling 350 Calls using a single Server and please note the Server will used to transferring the call only. Other Servers will handle gateway Negotiation and Billing. This server will SIMPLY be a Gateway. Please let me know if this configuration too high or too low. If anybody has better solution please let me know that as well. Thank you, waiting eagerly for a response. Imran M Yousuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Dear Users, I am fairly new to Digium and Asterisk. I wanted to know that if I use the Digium product THREE Digium Wildcard TE412Ps (Quad E1 Card) how many calls can I handle simultaneously. I want to use the cards with the following Configurations: Intel® Xeon 3.00GHz/800MHz, 2M Processor 1GB PC2-3200 DDR2 SDRAM (2x512MB + 2X1GB) memory Integrated Dual Channel Ultra320 SCSI Adapter NC7781 Single Port PCI-X embedded NIC Hot plug drive cage - Ultra3 (6X1) High Speed IDE CD-ROM Drive 72.8GB Pluggable Ultra320 SCSI 15,000 rpm (1) Universal Hard Drive Asterisk Business Edition 3 X TE412P I have a requirement of handling 350 Calls using a single Server and please note the Server will used to transferring the call only. Other Servers will handle gateway Negotiation and Billing. This server will SIMPLY be a Gateway. Please let me know if this configuration too high or too low. If anybody has better solution please let me know that as well. Thank you, waiting eagerly for a response. Imran M Yousuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Server
Hi Dear Users, I am new to Asterisk and had a query which is probably primitive. I wanted to know whether I can use the Digium Hardware and receive and establish connection to a host SIP Server which is totally a different platform. Let me explain - Usually there is a E1-VoIP gateway (independent Hardware) connecting to a Server/Client via LAN. In my case, SIP Server. Now what I want is that Digium PCI Hardware and the SIP Server will be the same PC and I Want the PCI Hardware to act as the gateway. Therefore my question in particular is: That is can I configure the device to talk to the Server in SIP protocol directly?-- Imran M Yousuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] my asterisk crashed
On 5/3/06, Goke Aruna [EMAIL PROTECTED] wrote: ... #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46 OUTBOUND_GROUP) at pbx.c:5904 #2 0xf5bbe1e4 in dial_exec_full (chan=0xa281820, data=0x0, peerflags=0xf469fee8) at app_dial.c:964 #3 0xf5bc23ed in dial_exec (chan=0x0, data=0x1) at app_dial.c:1601 This indicates a corrupted global variable list, This issue was fixed in 1.2.6, please upgrade. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error : ast_readaudio_callback: Failed to write frame
On 4/30/06, Hatami Nugraha [EMAIL PROTECTED] wrote: Hi all, I always get this error message after I hangup a call, what does it mean ? WARNING[8957]: file.c:583 ast_readaudio_callback: Failed to write frame This means you hungup while asterisk was trying to play a file to you. It should be of no concern as long as it does not happen during a call. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speeding up the dial of DTMF's in SIP channel
Please Ignore if you cannot edit the code.
You will have to modify app_dial.c in apps directory.
Look for code that calls ast_dtmf_stream(chan, ..., timeout)
The last parameter is the inter digit timeout, it can be set to as low
as 1 (1 millisec) a value of 0 it will default to 100millisecs.
The solution is to add an option to dial application for the timeout
which defaults to the current value(250ms) in app_dial which will
provide for custom timeouts through the dialplan.
Also note that too small timeouts like below 100ms will mess up inband
dtmf tones for example in some zap channels.
Imran
On 3/15/06, Álvaro Palma [EMAIL PROTECTED] wrote:
I'm dialing DTMF's in a SIP channel using the options:
[sip.conf]
dmtfmode=info
[extensions.conf]
exten = _XXX,1,Dial(SIP/gateway,,D(${EXTEN}))
(this is a custom SIP gateway, which receives the DTMF's sent from
softphones through Asterisk, and based on them, build the destination
PSTN number).
My problem is that Dial send the DTMF's to the SIP/gateway user at a
rate of about 1 DTMF each 300ms. I'd like to know if it's possible to
speed up that rate, or even, if it's possible to send the entire
extension as a single DTMF string.
Does anybody has a clue about how to do this? I was looking the options
for the Dial command, and nothing like that appears on it.
Thanks a lot for your help.
--
Atly.
Álvaro Palma
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Re: [Asterisk-Users] TE411P VPM
Use: modprobe wct4xxp vpmsupport=0 On 3/1/06, Aaron Daniel [EMAIL PROTECTED] wrote: Does anyone know how to disable the VPM in software rather than removing the card altogether? The canceler isn't working as well as the software cancelers were. Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] lists problem, Gmail????????
I have experienced similar problems using gmail. Gmail certainly had some problems with emails from asterisk lists. I donot know if it was only restricted to asterisk lists. As not all emails were being delayed (or dropped), some of you might be under the impression that theres no problem. Please compare your emails with the list archives to be sure you didnt miss something important. Also, the problems seems to have gone away this week. Regards Imran On 2/13/06, Joseph Tanner [EMAIL PROTECTED] wrote: May be some truth to it though :( Personally I use gmail, but use a different email address that is forwarded to my gmail account. With this setup, I haven't had any issues. I use gmail because it's easily accessible from any PC, and I like how it groups conversations (probably why you see a lot of gmail addresses signed up on mailing lists). Joseph Tanner On 2/13/06, Olivier.taylor [EMAIL PROTECTED] wrote: Pfff, What for an answer :( I use gmail and have no problems. Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Martin Joseph Envoyé : lundi 13 février 2006 20:36 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] lists problem, Gmail On Feb 13, 2006, at 2:45 AM, Simone Cittadini wrote: C F ha scritto: Am I the only one having trouble with this list? Since the begining of the week I have not been receiving mail from the list like I used to, is this a gmail problem? or is it subscription problem? or is something wrong with the list? anybody else using gmail having any problems? Yes, I'm also getting some lag sometimes, one or two days without receiving mails get a real mail server and it works great! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo cancel from telco
here is a little explanation: End user (You) - Your Telco -- Carrier 1 --- Carrier 2 Carrier 3 --- Carrier 4(PTT) --- Far End User So basically, the Echo cancelling work backwards usually cancellation for you would be done by Carrier 4, 3, 2, 1, or your Telco in that order and echo for the Far End User would be done by Your Telco, Carrier 1, 2, 3, or 4 in that order. Why in that order? AFAIK, the order is exactly the opposite, and if the user is experiencing echo on the sip phone, its most likely that the other end is the source of echo, which should be cancelled by the telco because its is nearer to the source of echo than the sip phone gateway. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo cancel from telco
On 2/7/06, Imran Ahmed [EMAIL PROTECTED] wrote: here is a little explanation: End user (You) - Your Telco -- Carrier 1 --- Carrier 2 Carrier 3 --- Carrier 4(PTT) --- Far End User So basically, the Echo cancelling work backwards usually cancellation for you would be done by Carrier 4, 3, 2, 1, or your Telco in that order and echo for the Far End User would be done by Your Telco, Carrier 1, 2, 3, or 4 in that order. Why in that order? AFAIK, the order is exactly the opposite, and if the user is experiencing echo on the sip phone, its most likely that the other end is the source of echo, which should be cancelled by the telco because its is nearer to the source of echo than the sip phone gateway. Never mind! I took the wording in a wrong way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rtp packets being dropped
AFAIK asterisk does not drop the packets, it just turns them into silence if it detects a dtmf. On 2/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello Friends, I am experiencing a problem. The rtp packets which detect dtmf from inband are being dropped. I have tried a priority ip address which allows voip packets first but it didnt work out. Asterisk is dropping only dtmf packets. I am using Sip protocol. Is there any way in asterisk whereby I can detect the dropped packets or enable their queueing or buffering? Please help, I am running out of ideas. Thanking you all. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Sweat saves blood, blood saves lives, and brains saves both. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
Step 3 The Iax client heve to send some other DTMF to the IVR. How is the IVR still involved if the call has been transferred into a conference room? The IVR records the conversation between the other partecipant to the conference and wait '#' to stop recording and a '1' to save the file. may or may not work, try at your own risk: 1) Use a sip soft phone and set the dtmf mode = inband. 2) In asterisk set the dtmf mode for that soft phone to be rfc2833 or info. (this is done so that asterisk ignores the inband dtmf on the sip channel). 3) Design your dialplan such that asterisk should not depend on dtmf from the sip call. ex: exten xxx, 1, dial(zap/g/client_number) //on answer directed to conference room exten xxx, 2, dial(zap/g/ivr_number) //on answer directed to conference room. exten xxx, 3, meetme(conference room) once the sip call is in the conference then the ivr will detect dtmf from the audio data. Note that before the sip call is in a conference dtmf will not be detectable by the ivr or asterisk, and Ofcourse, this is not tested and only a test can confirm if it works. drawbacks: dtmf will not be available to ivr until your call is in conference. asterisk will never see any dtmf (which should be okay in this specific case). dtmf tones are not squelched so the other user in the conference will hear dtmf tones. Imran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
AFAIK there's no DTMF option in IAX2... IAX always sends DTMF inline, eliminating the confusion often found with SIP. http://www.voip-info.org/wiki-IAX Even though no IAX client supports inband dtmf, An IAX client can send inband dtmf which would have corrected your problem. The problem here is with the meetme application when dealing with non zaptel channels it does not have a mechanism to enable dtmf to pass through the conference unless dtmf is inband (i.e. part of the audio stream). The following are the solutions a) Use a SIP phone with inband dtmf (No guarantee this will work either) b) Modify meetme to broadcast dtmf to all channels in conference( All channels will work in this case). Imran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
On 2/1/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Imran Ahmed wrote: Even though no IAX client supports inband dtmf, An IAX client can send inband dtmf which would have corrected your problem. No, it won't. No IAX2 client will start a DSP to listen for inband DTMF, because IAX2 is defined to always send out-of-band DTMF. At best, if the receiving IAX2 system is just passing the audio along to another protocol that does support inband DTMF, then sending it in the audio stream would work. If the application receiving the DTMF is on the other IAX2 end, though (like MeetMe in this case), then it will never 'see' the DTMF, because Asterisk will not look in the audio stream for DTMF. I agree, but the other ends of the conference were zap channels in this case, at least that is what I figured by the first email. Imran. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. I donot know if IaxComm has inband dtmf mode available, if so enable it and see if it works. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pri Gateway Hardware
You donot need multiple asterisk boxes for a single t1. A single p4 box should be helpful, you can use digiums te110p pci card for a single pri line into the box. The same box could also be on another network dealing with SIP. On 1/9/06, Carlos Alperin [EMAIL PROTECTED] wrote: All that you need is at least two boxes: 1 is going to be the PRI Asterisk box, which interfaces with the outside world. Also has to be able to communicate via SIP or IAX with the second box. 2 is the real Asterisk pbx with all the extensions, and the pbx features. Also has to be able to communicate via SIP or IAX with the first box. Done. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johnathan Falk Sent: Monday, January 09, 2006 4:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Pri Gateway Hardware Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesn't have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost and the quality? Thanks Johnathan Falk Network Administrator Clinton Community Schools ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.2.1 and mixmonitor problem
I think the broken pipe issue is related with the mpg123 player, try disabling moh and see if it behaves the same way On 12/19/05, Maximiliano J. Goldsmid [EMAIL PROTECTED] wrote: I have the same problem !! :-( 2005/12/18, Mohammad Shokuie [EMAIL PROTECTED]: Hi there, Any one confronted a crash in asterisk when using mixmonitor app. When i'm using the mixmonitor app on a briged call as soon as the called party hangs up the call asterisk crashes and the process terminates with following error message : Segmentation fault. Ouch .. error while writing audion data :: broken pipe but when the calling party hangs up, everything is smooth. Anyone has any idea on this issue? TIA. M. Shokuie Nia _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] number of users in a meetme conference
Hi All, I want to know what is the maximum number of users allowed in a single meetme conference. How far is this number practically feasible Thanks Imran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming context problem
Hello All, Can anyone u pls tell me the context pattern i need to add on sip.conf and extension.conf for incoming calls ... the senerio is i have a provider who routes a UK DID to my IP previously i was using ATA186 and calls were coming on ATA186 via sip and phone was connected to port 1 .. i didn`t had to do anything.. i want to use asterisk to attend the call and forward to a extension. how shld write the context for sip and extension.conf ? best wishes Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming context problem
Hello All, Can anyone u pls tell me the context pattern i need to add on sip.conf and extension.conf for incoming calls ... the senerio is i have a provider who routes a UK DID to my IP previously i was using ATA186 and calls were coming on ATA186 via sip and phone was connected to port 1 .. i didn`t had to do anything.. i want to use asterisk to attend the call and forward to a extension. how shld write the context for sip and extension.conf ? best wishes Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip calling errors
Hi Pros,
I`m new to Asterisk Getting following errors on my * :
-- Executing Dial(SIP/1000-ee7c, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Apr 28 21:06:09 WARNING[2268]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/venus-e8ba(2) to SIP/1000-ee7c(256)
Apr 28 21:06:09 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
Apr 28 21:06:09 NOTICE[2268]: channel.c:1724 ast_set_read_format:
Unable to find a path from g729 to gsm
Apr 28 21:06:09 NOTICE[2268]: channel.c:1691 ast_set_write_format:
Unable to find a path from gsm to g729
-- SIP/venus-e8ba is making progress passing it to SIP/1000-ee7c
RFC3389: 1 bytes, level 256...
Apr 28 21:06:10 NOTICE[2268]: rtp.c:298 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
Apr 28 21:06:13 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
-- SIP/venus-e8ba answered SIP/1000-ee7c
-- Attempting native bridge of SIP/1000-ee7c and SIP/venus-e8ba
Apr 28 21:06:21 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)
Apr 28 21:06:22 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)onse)
My SIP.CONF
[general]
port = 5060
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=g723.1
allow=alaw
allow=ulaw
allow=gsm
allow=g729
[venus]
type=friend
context=sip-dial
host=2.2.2.2
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=g729
insecure=very
dtmfmode=info
tos=0x18
[]
type=friend
host=dynamic
username=
secret=imran
dtmf=inband
context=internal
dtmfmode=rfc2833
[1000]
type=friend
username=1000
;secret=password1
host=dynamic
allow=g729
allow=g723.1
context=internal
dtmfmode=rfc2833
=
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/
PHONE2=SIP/1000
PHONE3=SIP/1001
[internal]
include = local-sip
[local-sip]
exten = ,1,Dial(${PHONE1},40,t)
exten = ,2,Hangup
exten = 1000,1,Dial(${PHONE2},40,t)
exten = 1000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
exten = _00.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00.,2,Hangup
Venus is my SIP provider (sorry u might hav guessed already)
1000 and 1001 belongs to my AudioCodec MP108 8 FXS SIP device and
is my softphone SJphone, i can dial soft to hard and vise versa, i can
call to US number thru my SIP provider using my Sjphone (crapy sound)
but when i try to dial from MP108 i get the above errors i mentioned.
MP108 have preloaded codec i.e. g729 and g723.1, my provider supports
g729 and g723.1
please can anyone help me ?
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[Asterisk-Users] Fwd: Sip calling errors
-- Forwarded message --
From: iMRAN [EMAIL PROTECTED]
Date: May 1, 2005 12:16 PM
Subject: Sip calling errors
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com, Alexander Scheerschmidt
[EMAIL PROTECTED]
Hi Pros,
I`m new to Asterisk Getting following errors on my * :
-- Executing Dial(SIP/1000-ee7c, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Apr 28 21:06:09 WARNING[2268]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/venus-e8ba(2) to SIP/1000-ee7c(256)
Apr 28 21:06:09 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
Apr 28 21:06:09 NOTICE[2268]: channel.c:1724 ast_set_read_format:
Unable to find a path from g729 to gsm
Apr 28 21:06:09 NOTICE[2268]: channel.c:1691 ast_set_write_format:
Unable to find a path from gsm to g729
-- SIP/venus-e8ba is making progress passing it to SIP/1000-ee7c
RFC3389: 1 bytes, level 256...
Apr 28 21:06:10 NOTICE[2268]: rtp.c:298 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
Apr 28 21:06:13 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
-- SIP/venus-e8ba answered SIP/1000-ee7c
-- Attempting native bridge of SIP/1000-ee7c and SIP/venus-e8ba
Apr 28 21:06:21 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)
Apr 28 21:06:22 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)onse)
My SIP.CONF
[general]
port = 5060
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=g723.1
allow=alaw
allow=ulaw
allow=gsm
allow=g729
[venus]
type=friend
context=sip-dial
host=2.2.2.2
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=g729
insecure=very
dtmfmode=info
tos=0x18
[]
type=friend
host=dynamic
username=
secret=imran
dtmf=inband
context=internal
dtmfmode=rfc2833
[1000]
type=friend
username=1000
;secret=password1
host=dynamic
allow=g729
allow=g723.1
context=internal
dtmfmode=rfc2833
=
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/
PHONE2=SIP/1000
PHONE3=SIP/1001
[internal]
include = local-sip
[local-sip]
exten = ,1,Dial(${PHONE1},40,t)
exten = ,2,Hangup
exten = 1000,1,Dial(${PHONE2},40,t)
exten = 1000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
exten = _00.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00.,2,Hangup
Venus is my SIP provider (sorry u might hav guessed already)
1000 and 1001 belongs to my AudioCodec MP108 8 FXS SIP device and
is my softphone SJphone, i can dial soft to hard and vise versa, i can
call to US number thru my SIP provider using my Sjphone (crapy sound)
but when i try to dial from MP108 i get the above errors i mentioned.
MP108 have preloaded codec i.e. g729 and g723.1, my provider supports
g729 and g723.1
please can anyone help me ?
___
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[Asterisk-Users] SIP Errors from MP108 please help - confs included
Hi Pros,
I`m new to Asterisk Getting following errors on my * :
-- Executing Dial(SIP/1000-ee7c, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Apr 28 21:06:09 WARNING[2268]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/venus-e8ba(2) to SIP/1000-ee7c(256)
Apr 28 21:06:09 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
Apr 28 21:06:09 NOTICE[2268]: channel.c:1724 ast_set_read_format:
Unable to find a path from g729 to gsm
Apr 28 21:06:09 NOTICE[2268]: channel.c:1691 ast_set_write_format:
Unable to find a path from gsm to g729
-- SIP/venus-e8ba is making progress passing it to SIP/1000-ee7c
RFC3389: 1 bytes, level 256...
Apr 28 21:06:10 NOTICE[2268]: rtp.c:298 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
Apr 28 21:06:13 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
-- SIP/venus-e8ba answered SIP/1000-ee7c
-- Attempting native bridge of SIP/1000-ee7c and SIP/venus-e8ba
Apr 28 21:06:21 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)
Apr 28 21:06:22 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)onse)
My SIP.CONF
[general]
port = 5060
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=g723.1
allow=alaw
allow=ulaw
allow=gsm
allow=g729
[venus]
type=friend
context=sip-dial
host=2.2.2.2
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=g729
insecure=very
dtmfmode=info
tos=0x18
[]
type=friend
host=dynamic
username=
secret=imran
dtmf=inband
context=internal
dtmfmode=rfc2833
[1000]
type=friend
username=1000
;secret=password1
host=dynamic
allow=g729
allow=g723.1
context=internal
dtmfmode=rfc2833
=
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/
PHONE2=SIP/1000
PHONE3=SIP/1001
[internal]
include = local-sip
[local-sip]
exten = ,1,Dial(${PHONE1},40,t)
exten = ,2,Hangup
exten = 1000,1,Dial(${PHONE2},40,t)
exten = 1000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
exten = _00.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00.,2,Hangup
Venus is my SIP provider (sorry u might hav guessed already)
1000 and 1001 belongs to my AudioCodec MP108 8 FXS SIP device and
is my softphone SJphone, i can dial soft to hard and vise versa, i can
call to US number thru my SIP provider using my Sjphone (crapy sound)
but when i try to dial from MP108 i get the above errors i mentioned.
MP108 have preloaded codec i.e. g729 and g723.1, my provider supports
g729 and g723.1
please can anyone help me ?
___
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[Asterisk-Users] SIP calling Error from MP108 please help - confs included
Hi Pros,
I`m new to Asterisk Getting following errors on my * :
-- Executing Dial(SIP/1000-ee7c, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Apr 28 21:06:09 WARNING[2268]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/venus-e8ba(2) to SIP/1000-ee7c(256)
Apr 28 21:06:09 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
Apr 28 21:06:09 NOTICE[2268]: channel.c:1724 ast_set_read_format:
Unable to find a path from g729 to gsm
Apr 28 21:06:09 NOTICE[2268]: channel.c:1691 ast_set_write_format:
Unable to find a path from gsm to g729
-- SIP/venus-e8ba is making progress passing it to SIP/1000-ee7c
RFC3389: 1 bytes, level 256...
Apr 28 21:06:10 NOTICE[2268]: rtp.c:298 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
Apr 28 21:06:13 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
-- SIP/venus-e8ba answered SIP/1000-ee7c
-- Attempting native bridge of SIP/1000-ee7c and SIP/venus-e8ba
Apr 28 21:06:21 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)
Apr 28 21:06:22 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)onse)
My SIP.CONF
[general]
port = 5060
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=g723.1
allow=alaw
allow=ulaw
allow=gsm
allow=g729
[venus]
type=friend
context=sip-dial
host=2.2.2.2
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=g729
insecure=very
dtmfmode=info
tos=0x18
[]
type=friend
host=dynamic
username=
secret=imran
dtmf=inband
context=internal
dtmfmode=rfc2833
[1000]
type=friend
username=1000
;secret=password1
host=dynamic
allow=g729
allow=g723.1
context=internal
dtmfmode=rfc2833
=
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/
PHONE2=SIP/1000
PHONE3=SIP/1001
[internal]
include = local-sip
[local-sip]
exten = ,1,Dial(${PHONE1},40,t)
exten = ,2,Hangup
exten = 1000,1,Dial(${PHONE2},40,t)
exten = 1000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
exten = _00.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00.,2,Hangup
Venus is my SIP provider (sorry u might hav guessed already)
1000 and 1001 belongs to my AudioCodec MP108 8 FXS SIP device and
is my softphone SJphone, i can dial soft to hard and vise versa, i can
call to US number thru my SIP provider using my Sjphone (crapy sound)
but when i try to dial from MP108 i get the above errors i mentioned.
MP108 have preloaded codec i.e. g729 and g723.1, my provider supports
g729 and g723.1
please can anyone help me ?
___
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[Asterisk-Users] SIP calling Error from MP108 please help - confs included
Hi Pros,
I`m new to Asterisk Getting following errors on my * :
-- Executing Dial(SIP/1000-ee7c, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Apr 28 21:06:09 WARNING[2268]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/venus-e8ba(2) to SIP/1000-ee7c(256)
Apr 28 21:06:09 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
Apr 28 21:06:09 NOTICE[2268]: channel.c:1724 ast_set_read_format:
Unable to find a path from g729 to gsm
Apr 28 21:06:09 NOTICE[2268]: channel.c:1691 ast_set_write_format:
Unable to find a path from gsm to g729
-- SIP/venus-e8ba is making progress passing it to SIP/1000-ee7c
RFC3389: 1 bytes, level 256...
Apr 28 21:06:10 NOTICE[2268]: rtp.c:298 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
Apr 28 21:06:13 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
-- SIP/venus-e8ba answered SIP/1000-ee7c
-- Attempting native bridge of SIP/1000-ee7c and SIP/venus-e8ba
Apr 28 21:06:21 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)
Apr 28 21:06:22 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)onse)
My SIP.CONF
[general]
port = 5060
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=g723.1
allow=alaw
allow=ulaw
allow=gsm
allow=g729
[venus]
type=friend
context=sip-dial
host=2.2.2.2
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=g729
insecure=very
dtmfmode=info
tos=0x18
[]
type=friend
host=dynamic
username=
secret=imran
dtmf=inband
context=internal
dtmfmode=rfc2833
[1000]
type=friend
username=1000
;secret=password1
host=dynamic
allow=g729
allow=g723.1
context=internal
dtmfmode=rfc2833
=
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/
PHONE2=SIP/1000
PHONE3=SIP/1001
[internal]
include = local-sip
[local-sip]
exten = ,1,Dial(${PHONE1},40,t)
exten = ,2,Hangup
exten = 1000,1,Dial(${PHONE2},40,t)
exten = 1000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
exten = _00.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00.,2,Hangup
Venus is my SIP provider (sorry u might hav guessed already)
1000 and 1001 belongs to my AudioCodec MP108 8 FXS SIP device and
is my softphone SJphone, i can dial soft to hard and vise versa, i can
call to US number thru my SIP provider using my Sjphone (crapy sound)
but when i try to dial from MP108 i get the above errors i mentioned.
MP108 have preloaded codec i.e. g729 and g723.1, my provider supports
g729 and g723.1
please can anyone help me ?
___
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[Asterisk-Users] Route SIP calls to provider
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729
[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000
[international]
ignorepat = 88
exten= _1N1NXXNXX,1,Dial ???
[internal]
include = local-sip
[local-sip]
exten = 1000,1,Dial(${PHONE1},40,t)
exten = 1000,2,Hangup
exten = 2000,1,Dial(${PHONE2},40,t)
exten = 2000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
i want user to dial 88 and they will get a tone and dial US or UK
number from local-sip context.
the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International
context in extension.conf also what i shld do in sip.conf..
my audiocodec supports g729 and g723 codec so do i need to aqquire
license for G729 from digium, if yes then why?
last if possible can you also please tell me wht i need to add on my
context so user can while in calling put the call on hold and transfer
to another sip phone.
I thankyou all for reading this mail and i hope someone will be kind
enough to help.
Best regards,
Imran
___
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[Asterisk-Users] Newbie - VoIP route SIP calls to provider
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729
[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000
[international]
ignorepat = 88
exten= _1N1NXXNXX,1,Dial ???
[internal]
include = local-sip
[local-sip]
exten = 1000,1,Dial(${PHONE1},40,t)
exten = 1000,2,Hangup
exten = 2000,1,Dial(${PHONE2},40,t)
exten = 2000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
i want user to dial 88 and they will get a tone and dial US or UK
number from local-sip context.
the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International
context in extension.conf also what i shld do in sip.conf..
my audiocodec supports g729 and g723 codec so do i need to aqquire
license for G729 from digium, if yes then why?
last if possible can you also please tell me wht i need to add on my
context so user can while in calling put the call on hold and transfer
to another sip phone.
I thankyou all for reading this mail and i hope someone will be kind
enough to help.
Best regards,
Imran
___
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[Asterisk-Users] Newbie - VoIP route SIP calls to provider
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729
[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000
[international]
ignorepat = 88
exten= _1N1NXXNXX,1,Dial ???
[internal]
include = local-sip
[local-sip]
exten = 1000,1,Dial(${PHONE1},40,t)
exten = 1000,2,Hangup
exten = 2000,1,Dial(${PHONE2},40,t)
exten = 2000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
i want user to dial 88 and they will get a tone and dial US or UK
number from local-sip context.
the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International
context in extension.conf also what i shld do in sip.conf..
my audiocodec supports g729 and g723 codec so do i need to aqquire
license for G729 from digium, if yes then why?
last if possible can you also please tell me wht i need to add on my
context so user can while in calling put the call on hold and transfer
to another sip phone.
I thankyou all for reading this mail and i hope someone will be kind
enough to help.
Best regards,
Imran
___
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[Asterisk-Users] VoIP route SIP calls to provider
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729
[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000
[international]
ignorepat = 88
exten= _1N1NXXNXX,1,Dial ???
[internal]
include = local-sip
[local-sip]
exten = 1000,1,Dial(${PHONE1},40,t)
exten = 1000,2,Hangup
exten = 2000,1,Dial(${PHONE2},40,t)
exten = 2000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
i want user to dial 88 and they will get a tone and dial US or UK
number from local-sip context.
the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International
context in extension.conf also what i shld do in sip.conf..
my audiocodec supports g729 and g723 codec so do i need to aqquire
license for G729 from digium, if yes then why?
last if possible can you also please tell me wht i need to add on my
context so user can while in calling put the call on hold and transfer
to another sip phone.
I thankyou all for reading this mail and i hope someone will be kind
enough to help.
Best regards,
Imran
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Want to use Asterisk instead of existing Meridian Norstar system ... need some help
Hello: I want to replace our Meridian system (6 incoming lines) with * for obvious reasons such as Voip but I am not sure how to go about this. I have 6 incoming lines with a rollover so that only one main number is published. I understand that I will need 6 FXO ports coming into Asterisk but I am not sure how to handle connections going out of * (the connections that my existing phones will use) how many FXS ports will I need Can I use my existing Meridian phones (6 line ones) Do I need 1 FXS port for each extensions in the offices (there are currently 12 meridian phones) How should the extension jacks in the offices be wired I guess this will depend on the answer to # 2 above. Any help will be greatly appreciated. Best regards Imran -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.16 - Release Date: 4/18/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Softphone for testing with Asterisk
Hi Ray I`m using SJphone softphones with my * , working fine.. Imran On Apr 7, 2005 8:41 AM, raymond [EMAIL PROTECTED] wrote: Hi all, I had just set up my asterisk server. Can anybody know that is there any sip softphone for testing with asterisk? (I had download some from internet but I think all are preconfig to certain server). Cheers. Raymond ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fedora 3
Hi, I`ve installed on FC-3 last month and its working gr8... no probs so far Imran On Apr 6, 2005 2:38 PM, Altus Snyman [EMAIL PROTECTED] wrote: Good day all I have a Fedora core 3 installation Is there any hassles with asterisk? Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio codec MP108 please help
hi all, can any 1 pls tell me the context i shld add on sip.conf for Audiocodec MP108 8 fxs please. i want to add 2 phone on MP108 port assign extention and dial each other, can`t get a dialtone only busy signal. Thnx ppls Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What is ZAP ? newbie question sorry
Hi Pros, Please advice whats the purpose of ZAP, if i have softphones and ATA 186 with PSTN trunk, wht ZAP will do ? do i zap to route calls internal softphone to softphones ? thnx a lot Ronny ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio codec MP108
hi all, can any 1 pls tell me the context i shld add on sip.conf for Audiocodec MP108 8 fxs please. can`t get a dialtone only busy signal. Thnx ppls Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] lost newbie requesting help for Asterisk Implementation
Hi friends, i`m totally a newbie on VoIP let alone asterisk. I`m very much interested in learning asterisk to deploy on my small Call Center, we have 2 audioCodec MP-108 8 fxs port SIP device and 6 A800 H323 analog quintums. I installed fresh asterisk with samples, might b peice of cake for u all but for its hard for me even to get 2 softphone installed on 2 pcs to comunicate with each other using my Asterisk. 1st of all i want to learn how and wht i need to configure to use my 2 softphone comunicate on the internal network. 2nd i desparately need to figureout to configure MP108 and route out 800 number to land on asterisk and forward to 1 port of mp108 and use the 7 port which i will assign to my seniors to so that 2 VIPs can dial US and rest can dial UK numbers. we have overseas call provider and currently we r calling from gnugk openh323 server to place calls from quintums. I am planning to aquire E1 card after i have successfully worked on the above senerio, i have an iDCS and planning to configure it with Asterisk for whole org. I`ld very much apprecite if any 1 could guide me to implement asterisk on my network please. Ronny ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP H323 gateway
Hi pros, Newbie to asterisk, need some help. My existing senerio is we have 6 analog quintums and 1 digital H323, and our gatekeeper is gnugk openh323 located in US. Our business is Call Center and our method of dial is using prefix and gateway IP provided my Carrier. I also brought two AudioCodes MP108 8 FXS gateways, as our gateway runs on h323 my friend suggested to go for Asterisk. If I'm not mistaken Asterisk can entertain both H323 and SIP so I need to configure Asterisk as SIP and H323 gatekeeper to take calls and route to our International Carrier, I installed Asterisk on Fedora core 3, installation was successful but when I start asterisk with vvvc after 5-10 mins the box freezes, don`t know why, and it only happens whn I start asterisk, now i`m installing RH 9. The digital quintum have 4 E1 ports and we have brought iDCS with dialogic card installed, is thr anyway I could use Asterisk for this ? My question what files I need to modify on asterisk for SIP and H323 working and place calls. Can any 1 send a sample config for SIP and h323 please? Best regards, Mohd. Imran Kamal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What is the best and easiest flavor to be used with Asterisk.
Could anyone please advise me on the best flavor of Linux on which Asterisk is easiest to install. I am currently using RH8.0, everything over the IP works fine but when I want to call a physical line I can only have conversation for about 3 sec and everything freezes after that. I have to hard reset the machine to bring it back up. Any suggestions will be greatly appreciated. Thanks Imran Sadiq Systems Engineer Tel: +64 9 377 8282 World Class Support for any business Fax: +64 9 377 7900 with between 7 and 70 computers. Mob: 027 286 9269 LANcom Technology Limited: 25 Union St, Auckland, New Zealand image001.jpgimage002.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line drops after 5-10 seconds
Hi, I got a weird problem. If I call someone through fxo line, The bell rings, then I can only talk for around 2-3 seconds. After that I cannot hear anything i.e. the line drops and the asterisk server crashes i.e. it freezes. Does anyone have any clues Thanks, Imran Sadiq Systems Engineer Tel: +64 9 377 8282 World Class Support for any business Fax: +64 9 377 7900 with between 7 and 70 computers. Mob: 027 286 9269 LANcom Technology Limited: 25 Union St, Auckland, New Zealand image001.jpgimage002.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Just saw your [Asterisk] xJack Segfault in Asterisk
Well, I had to compile the Mepis source and install it again. It did compile the Zaptel drivers but then is started giving other problems. Like it would not instalel ztdummy. Therefore I have given up on Mepis and downloading Red Hat 8.0 now. I will install Asterisk on that now. Thanks for the interest. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cmould Sent: Tuesday, 4 January 2005 1:18 a.m. To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Just saw your [Asterisk] xJack Segfault in Asterisk Hi: Just saw your post while trying to solve a similar asterisk problem. Did not see any responses. Was your problem solved and what was the solution? Carey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems installing Zaptel
Hi, I am new to asterisk. I have downloaded Asterisk and Zaptel from the cvs root. I am installing them on Mepis with linux-2.6.7 Whenever I try to do make in the zaptel directory, I get the following errors. make C /lib/modules/`uname r` /build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.7` make[1]: *** No rule to make target `modules`. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.7` make: *** [linux26] Error 2 I found some help on that which instructs to make a symbolic link I did the following. ln s /usr/src/linux-2.6.7 - /lib/modules/build/linux-2.6.7It still gives me the same error. Does anyone have any suggestions?Thanks in advance.Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems installing Zaptel
Kristian, Thanks for that. It still gives me the same error. I have also tried make linux26 in the zaptel directory and it gives me the exact same error. I also tried make linux-2.6.7 and that says *** No rule to make target `linux-2.6.7' Any suggesstions? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Wednesday, 22 December 2004 10:33 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problems installing Zaptel Imran Sadiq wrote: Hi, I am new to asterisk. I have downloaded Asterisk and Zaptel from the cvs root. I am installing them on Mepis with linux-2.6.7 Whenever I try to do make in the zaptel directory, I get the following errors. make -C /lib/modules/`uname -r` /build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.7` make[1]: *** No rule to make target `modules`. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.7` make: *** [linux26] Error 2 I found some help on that which instructs to make a symbolic link I did the following. ln -s /usr/src/linux-2.6.7 - //lib/modules/build/linux-2.6.7/ / / It still gives me the same error. Does anyone have any suggestions? Thanks in advance. Imran Imran, Try this: ln -s /usr/src/linux-2.6.7 /usr/src/linux-2.6 and make linux26 in the Zaptel directory. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems installing Zaptel
Yes you are right. I installed Mepis straight from their CD. It does not look like there is any Makefile in the directory. I think it probably did not copy the source files. Please forgive my ignorance in Linux. I am pretty new to this stuff. What would be the easiest way out of this situation? Any help would be appreciated. Thanks, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Fineberg Sent: Wednesday, 22 December 2004 4:56 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problems installing Zaptel Imran Sadiq wrote: Kristian,Thanks for that.It still gives me the same error.I have also tried make linux26 in the zaptel directory and it gives methe exact same error.I also tried make linux-2.6.7 and that says *** No rule to maketarget `linux-2.6.7' Any suggesstions? Are you sure you have the kernel source installed and configured? The make target for 2.6.7 is linux26 as per the README.Linux26 so the only thought is there is no Makefile in /usr/src/linux-2.6.7. Thanks-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of KristianKielhofnerSent: Wednesday, 22 December 2004 10:33 a.m.To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Problems installing ZaptelImran Sadiq wrote: Hi, I am new to asterisk.I have downloaded Asterisk and Zaptel from the cvs root.I am installing them on Mepis with linux-2.6.7 Whenever I try to do make in the zaptel directory, I get the following errors. make -C /lib/modules/`uname -r` /build SUBDIRS=/usr/src/zaptel modulesmake[1]: Entering directory `/usr/src/linux-2.6.7`make[1]: *** No rule to make target `modules`. Stop.make[1]: Leaving directory `/usr/src/linux-2.6.7`make: *** [linux26] Error 2 I found some help on that which instructs to make a symbolic link I did the following. ln -s /usr/src/linux-2.6.7 - //lib/modules/build/linux-2.6.7// /It still gives me the same error. Does anyone have any suggestions? Thanks in advance. Imran Imran, Try this:ln -s /usr/src/linux-2.6.7 /usr/src/linux-2.6andmake linux26 in the Zaptel directory.--Kristian Kielhofner___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] digitnetworks card issues?
Hi,
I've purchased two x100p clones, and when I try accessing a line
from asterisk with something like this:
exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
(is that only supposed to put you on channel 2 or actually dial the #
for you?)
but I first hear noise, then a dial tone, but as soon as I start dialing
numbers I get feedback and noise, and the call doesn't go through.
Any suggestions?
Thanks,
Imran
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Re: [Asterisk-Users] digitnetworks card issues?
Didn't want to start a flamewar here... but anyway, could the issue be
that both fxo cards are on IRQ 11? How do I even change that?
Thanks
William Suffill wrote:
Digitnetworks is profiting off the cards so they should support them.
If it wasn't for Digium there wouldn't be Asterisk anyway. So doesn't
that make it better to support the primary company for software that
many of you use every day at home and work?
On Fri, 3 Sep 2004 08:40:59 +0100, Kannaiyan Natesan
[EMAIL PROTECTED] wrote:
Does it mean that we cannot talk about Cisco or other FXS products since
IAXy is released??
I hope this list for every member who uses asterisk not Digium's products
users alone.
- Original Message -
From: Jay Milk [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Friday, September 03, 2004 8:09 AM
Subject: RE: [Asterisk-Users] digitnetworks card issues?
Have you contacted digitnetworks for support? This list is owned and
maintained by Digium, who already gave you Asterisk for free. Probably
not the best forum to ask for support for a competitive product here.
-Original Message-
From: Imran Akbar [mailto:[EMAIL PROTECTED]
Sent: Friday, September 03, 2004 1:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] digitnetworks card issues?
Hi,
I've purchased two x100p clones, and when I try accessing a line
from asterisk with something like this:
exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
(is that only supposed to put you on channel 2 or actually dial the #
for you?)
but I first hear noise, then a dial tone, but as soon as I
start dialing
numbers I get feedback and noise, and the call doesn't go through.
Any suggestions?
Thanks,
Imran
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[Asterisk-Users] no dial tone when dialing out on vonage
Hi,
I'm trying to dial out on a vonage line connected to a zap channel
using stuff like:
exten = 200,1,Dial(Zap/2/${EXTEN})
but it doesn't work - when i dial in the extension, i can see on a phone
connected to the same line that it's gone active - but there's no
dialtone. also tried adding a wait period before accessing the line and
exten = _XX,1,Dial(Zap/2/${EXTEN})
to no avail.
what's goin on?
Thanks,
Imran
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[Asterisk-Users] line feedback, no dial tone
Hi, after following up on my previous email about zaptel x100p having trouble accessing a vonage dial tone, I think the problem is with feedback and noise on the line - any remedies for this? Thanks, Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] international caller id support
Hi, I was wondering if anyone knows where i can find out the standards used by telco's in different countries... and how to configure asterisk to support them. Secondly, whenever i try Dialing a zap channel, all i get is no sound on the phone source, and noise on the destination line. thanks Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transferring call to another line
Hi, I just got my zaptel fxo cards working, and I want to be able to have someone call in on one line and access the other - I guess what I want to do is transfer(exten), but that is only for extensions - not channels which is what I want i guess. I tried the Dial(Zap/2) but I think that's for ringing that line (fxs)? thanks Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions = s,1,Dial(Zap/2/number) noise
Hi, I'm trying to answer a call on one line and dial out a number on a zaptel x100p fxo, but all I get from the phone I'm dialing is silence after it is picked up, and on the line that's supposed to be dialed out itself, noise. Thanks, Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: zaptel configuration
Asalamualaikum Atif, i saw your guy's ad in spider magazine. sounds cool... yeah, i got asterisk to work, i had to build zaptel before asterisk. just trying to transfer from one line to another now... thanks Imran Atif Rasheed wrote: well Imran, I am not a guru of Asterisk, but I think my suggestions might work, Hi, I've been trying to get my zaptel x100p cards working for the past week now. this is what I've done: first, this should have been done with Zaptel, not asterisk install Zaptel, like this: make clean make linux26 (if kernel version is 2.6 or above, else do 'make' only) make install installed asterisk: make clean make linux 26 (for fedora core 2) make install and, this should have been done for Asterisk i.e. install Asterisk, like: make clean make make install installed zaptel: make clean make make install then do 'modprobe's', 'ztcfg -v', and then do 'asterisk -vvc' then check for errors, if any did a modprobe zaptel, and wcfxo got this in /var/log/messages: PCI: found IRQ 11 for device :00:0f.0 wcfxo: daa mode is 'FCC' found a wildcard fxo: wildcard x101p ... in zaptel.conf: fxsks=1-2 in zapata.conf: signalling = fxs_ks channel = 1 channel = 2 yet when i run asterisk, the zap show channels command doesn't work. in a previous thread they mentioned this is because some chan_zap.so file isn't loaded because of the zaptel installation. I was told I had to REINSTALL asterisk after the zaptel stuff, which again didn't do anything. How can this be so hard to even get installed? Thanks, Imran hope, it will work this time. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial/Zap doesn't work
Hi, I'm trying to dial in from one phone and give it access to another line (ie incoming on zap/1 and outgoing on zap/2)... how can I transfer the call from channel 1 and give it the dial tone on channel 2? I can use dial but that takes a phone number, which I want the user to be able to select. thanks Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel configuration
Hi, I've been trying to get my zaptel x100p cards working for the past week now. this is what I've done: installed asterisk: make clean make linux 26 (for fedora core 2) make install installed zaptel: make clean make make install did a modprobe zaptel, and wcfxo got this in /var/log/messages: PCI: found IRQ 11 for device :00:0f.0 wcfxo: daa mode is 'FCC' found a wildcard fxo: wildcard x101p ... in zaptel.conf: fxsks=1-2 in zapata.conf: signalling = fxs_ks channel = 1 channel = 2 yet when i run asterisk, the zap show channels command doesn't work. in a previous thread they mentioned this is because some chan_zap.so file isn't loaded because of the zaptel installation. I was told I had to REINSTALL asterisk after the zaptel stuff, which again didn't do anything. How can this be so hard to even get installed? Thanks, Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap show channels - no such command
dmesg: Zapata Telephony Interface Registered on major 196 ... pci: fOUND irq 11 for device 00 wcfxo: daa mode is 'FCC' Found a wildcard fxo: wildcard x101p PCI: found iraq 11 for device pci: sharing irq 11 with 0 wcfxo: DAA modeis 'FCC' Found a Wildcard FXO: Wildcard X101p that's for two FXO cards. Thanks el Flynn wrote: Imran Akbar wrote: edited the zaptel.conf, zapata.conf, extensions.conf to proper settings. added chan_zap.so to modules.conf, when asterisk starts up it can't find it. Why don't you post a snippet of the zaptel stuff as reported by dmesg? That may help. Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap show channels - no such command
wo, i have to rebuild asterisk after i install zaptel? where did that come from? let me try... thanks Imran Darryl Ross wrote: Hi Imran, I seem to have done the zaptel installation - what am I missing - i still don't have a chan_zap.so file? Did you rebuild Asterisk after installing Zaptel? chan_zap.so is built as part of the actually Asterisk build, AFAIK. added chan_zap.so to modules.conf, when asterisk starts up it can't find it. If you don't have Zaptel installed when you build Asterisk, it might not build chan_zap.so. On my system the asterisk modules are in /usr/lib/asterisk/modules/. Regards Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap show channels - no such command
Tried recompiling asterisk after the zaptel installation... still don't have a chan_zap.so file. help anyone? Thanks, Imran Imran Akbar wrote: wo, i have to rebuild asterisk after i install zaptel? where did that come from? let me try... thanks Imran Darryl Ross wrote: Hi Imran, I seem to have done the zaptel installation - what am I missing - i still don't have a chan_zap.so file? Did you rebuild Asterisk after installing Zaptel? chan_zap.so is built as part of the actually Asterisk build, AFAIK. added chan_zap.so to modules.conf, when asterisk starts up it can't find it. If you don't have Zaptel installed when you build Asterisk, it might not build chan_zap.so. On my system the asterisk modules are in /usr/lib/asterisk/modules/. Regards Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel installation
Hi, the problems i previously had with zap show channels seems to be from an incorrect zaptel installation which is why I don't have a chan_zap.so file. I compile and do a make clean, make, make install of zaptel and do my modprobe's, and I was told to reinstall asterisk after that. I do so however, but it makes no difference. any hints? thanks Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zap show channels - no such command
Hi, in response to a previous posting regarding getting the x100p to work, I was told to run zap show channels, but when i do i get no such command 'zap' There was a previous posting on this, but the guy never posted the solution. thanks, Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap show channels - no such command
Thanks Jeremy, but how exactly do I load chan_zap.so? I put it into my modules.conf, but when i run asterisk now it says it can't find it (loading module zap_chan.so failed). It doesn't seem to be on my system... thanks imran Jeremy McNamara wrote: Imran Akbar wrote: Hi, in response to a previous posting regarding getting the x100p to work, I was told to run zap show channels, but when i do i get no such command 'zap' There was a previous posting on this, but the guy never posted the solution. chan_zap.so is not loaded. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap show channels - no such command
to the best of my knowledge, i have, but i'm redoing it. i'm looking at the instructions at http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation is that the best guide? thanks Imran Jeremy McNamara wrote: Imran Akbar wrote: Thanks Jeremy, but how exactly do I load chan_zap.so? I put it into my modules.conf, but when i run asterisk now it says it can't find it (loading module zap_chan.so failed). It doesn't seem to be on my system... Have you compiled, installed and configured Zaptel? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap show channels - no such command
sorry, my bad. typo in the email, but it was correct in modules.conf. Im trying to reinstall the zaptel stuff, but i'm not seeing anything in var/log/messages after doing my modprobe's? Thanks Jon Radon wrote: It should be chan_zap.so not zap_chan.so. Imran Akbar wrote: Thanks Jeremy, but how exactly do I load chan_zap.so? I put it into my modules.conf, but when i run asterisk now it says it can't find it (loading module zap_chan.so failed). It doesn't seem to be on my system... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap show channels - no such command
Thanks, I seem to have done the zaptel installation - what am I missing - i still don't have a chan_zap.so file? in my zaptel directory: make clean make make install modprobe zaptel modprobe wcfxo got stuff in dmesg did a make config in the zaptel directory edited the zaptel.conf, zapata.conf, extensions.conf to proper settings. added chan_zap.so to modules.conf, when asterisk starts up it can't find it. Thanks, Imran el Flynn wrote: Imran Akbar wrote: sorry, my bad. typo in the email, but it was correct in modules.conf. Im trying to reinstall the zaptel stuff, but i'm not seeing anything in var/log/messages after doing my modprobe's? Thanks try running the dmesg command - the digium stuff appears there instead of /var/log/messages on my system. Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel config
Hi, Sorry, in my last mail I wrote wcfxs instead of what I actually used, wcfxo. I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxo, which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks context = line1 channel = 1 signalling = fxs_ks context = line1 channel = 2 zaptel.conf has: loadzone=us defaultzone=us fxsks=1-2 extensions.conf has: [line1] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,BackGround(demo-congrats) exten = s,5,BackGround(demo-instruct) [line2] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,BackGround(demo-congrats) exten = s,5,BackGround(demo-instruct) I have no idea why it's not working would appreciate any help Thanks, Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Local PBX
I am running Asterisk. I want to make my local PBX. I have Cisco ATA 186-I1. i want to connect two analog telephone connected to ATA 186 and make them extention to dial each other. how i can make it. Imme -- __ Sign-up for your own FREE Personalized E-mail at Mail.com http://www.mail.com/?sr=signup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1X1 PBX
I have Asterisk and Cisco ATA 186. How i can make small PBX. let me know the step and configuration made in conf files. Imme -- __ Sign-up for your own FREE Personalized E-mail at Mail.com http://www.mail.com/?sr=signup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
