Re: [asterisk-users] VoIP Termination in Japan

On 5/5/10, Adrian Marsh adrian.ma...@ubiquisys.com wrote: Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -- Sent from my

Re: [asterisk-users] VoIP Termination in Japan

On 5/5/10, Adrian Marsh adrian.ma...@ubiquisys.com wrote: Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -- Sent from my

Re: [asterisk-users] how does wrapuptime work in queue.conf

...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Andy Kuo wrote: Hi Barry, Thank you for the hint, but I forgot to mention that we have a few advertisements, and we want the callers to listen to only one at a time, and in a round robin or random order.  Using Playback() doesn't

Re: [asterisk-users] how does wrapuptime work in queue.conf

; and they are queued immediately, so no time is wasted. l. 2009/8/27 Andy Kuo aku...@gmail.com Hi Barry, Thank you for the hint, but I forgot to mention that we have a few advertisements, and we want the callers to listen to only one at a time, and in a round robin or random order.  Using Playback

[asterisk-users] how does wrapuptime work in queue.conf

Hi list, I'd like to have the callers to listen to the advertisement (music on hold) before the agents answer them. So, I have wrapuptime=10 in queue.conf, but the call still goes straight to the agents without delay. Here's my queue.conf: [general] persistentmembers = yes [738] musiconhold =

Re: [asterisk-users] how does wrapuptime work in queue.conf

On Thu, Aug 27, 2009 at 11:56 AM, Barry L. Klineblkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Andy Kuo wrote: Hi list, I'd like to have the callers to listen to the advertisement (music on hold) before the agents answer them.  So, I have wrapuptime=10

Re: [asterisk-users] help - How to send hangup command to call in progress.

Hi Singh, Have you tried soft hangup? Andy On Wed, Mar 25, 2009 at 4:38 PM, Singh Saimbhi singh.saim...@palm.com wrote: Hi, I want to send hangup command to the call which was logged in earlier via cli.  Lets say to '5aec0e7207b24c8e1bdb511a460f7...@callcentric.com Basically I want to

Re: [asterisk-users] remove queue call

Hi, Try CLI soft hangup Local. Andy On 8/27/08, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, I have the following queue and members. I found that there is a call stuck in the queue so other call can't enter the queue. I want to know whether we can remove the call (by CLI) to

Re: [asterisk-users] karaoke functionality

Hi, Why not use MixMonitor(), so you have a single file with the singer and the music? Thanks. Andy On 5/20/08, Sherwood McGowan [EMAIL PROTECTED] wrote: Arjan Kroon | Mobillion wrote: Hi, Is it possible top use a form of Karaoke Functionality? When a caller calls a

Re: [asterisk-users] zapata.conf: cannot set txgain lower than -6.3 ?

Hi Steve, I tried txgain as low as -18 without any problem, but I never tried anything with decimal points. Andy On 12/12/06, Steve Hsieh [EMAIL PROTECTED] wrote: Greetings everyone, I have a Digium TDM400P card with both an FXO and FXS module to connect to the phone company and to a

[asterisk-users] Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?

Hi, Has anyone seen this before? Sep 12 22:31:38 WARNING[17472]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? I found a bug tracker http://bugs.digium.com/view.php?id=6333 talking about this, but didn't really understand why it

[asterisk-users] strange problem with calls between MGCP and SIP clients(ATA's)

Hi, We have experience problems with calls between MGCP ATA's and SIP ATA's (Linksys PAP2-NA). A call from MGCP or SIP to the other connects normally and the conversation can usually last around 30 seconds and it becomes one-way audio. What I don't understand is how the calls can be set up and

Re: [asterisk-users] Asterisk t38passthrough

Hi Ricardo, On a 1.2.4 with the T.38 patch, I tried both t38pt_udptl = yes t38pt_rtp = yes t38pt_tcp = yes and t38pt_udptl = yes t38pt_rtp = no t38pt_tcp = no but still got ...chan_sip.c:3716 process_sdp: Unknown SDP media type in offer: image 5144 UDPTL t38 Warnings I tried it on Kapanga

Re: [asterisk-users] FAX questions

Hi Marco, I'm using T406P(with hardware EC) with a T1-PRI, and I'm having trouble sending fax out though SIP ATA in the same LAN subnet with the Asterisk box. I can send fax out using txfax in call file, but I did have to lower the rxgain and txgain. This is what I'm trying to do: Fax machine

Re: [asterisk-users] Re: need a pointer regarding scripting asterisk

Hi, Can you give a quick example on how to query an EXTERNAL database? Thank you. Andy On 7/29/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Jul 28, 2006 at 04:08:19PM -0500, shawn bright wrote: i would use a dial plan, but we are monitoring about 1200 units in the field, i thought a

Re: [Asterisk-Users] SOLVED: IAX jitter / clocking problem

Hi, I too would like to set a minimum jitterbuffer value, and that seems to mean that I need to use the old jitterbuffer implementation. Have you compared the 2 implementations? What are the advantages of using the new one and what are the disadvantages of using the old one? Thanks. Andy On

Re: [Asterisk-Users] call quality statistics?

try iax2 show netstats On 6/23/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote: Is it possible to set up some sort of call-quality statistics reporting/logging for IAX2 calls? Something that can keep track of dropped packet / jitter trends? (I know iax2 show channels shows this info for

Re: [Asterisk-Users] Sip t38 gateway tests

Hi Carlos, I missed the first part(s) of the conversation, but I think this is for Asterisk T.38 support. I tried the t.38 patch on Asterisk 1.2.4 about 2 months ago, but it still showed error messages when I tried t.38 fax on 2 t.38 enabled ATAs. (Grandstream HT286) Can I test T.38 with you?

Re: [Asterisk-Users] ast_sched_runq ran 281 scheduled tasks all at once

We get these messages too, but they don't seem to cause any problems. Are you connecting 2 * (with different versions) via IAX2? Are these messages only appear on the lower version one? I asked a similar question on the list, and the suggestion was to upgrade them to the same version. Hope this

Re: [Asterisk-Users] Steps to make trunked iax2

Hi, I'm connecting 2 Asterisk with IAX2. One is running 1.2.0, and the other is running 1.2.4 It has been working OK so far, except I get messages like these occassionally Apr 10 11:14:54 WARNING[13081]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 42 scheduled tasks all at

Re: [Asterisk-Users] What causes deadlock?

I sometimes get these WARNINGs too. I would like to know what causes it and how to avoid them too. Thanks. Andy On 4/5/06, Chuck Bunn [EMAIL PROTECTED] wrote: Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5

Re: [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P

Hi, We have Rhino channel bank and TE406, and are thinking of doing the same thing. What modem or modem pool are you using? Can Asterisk serve as an access server/gateway to the internet? Please share your experience. Thank you. Andy On 3/28/06, Don Pobanz [EMAIL PROTECTED] wrote: Nico

Re: [Asterisk-Users] Dial plan question - exclamtion mark

Try using . instead of ! _001800NXX _X. _X. is more like a match the rest instead of match all Hope this helps. Andy On 3/22/06, Mike Hammett [EMAIL PROTECTED] wrote: http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns says: ! wildcard,

Re: [Asterisk-Users] Unable to forward frame

Hi, I think there should be only one timing source, but you have 3 here... Zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 Not sure if this is causing the problem though. Andy On 3/15/06, James Sturges [EMAIL PROTECTED]

Re: [Asterisk-Users] Max retries exceeded to host...

What ATA's are you using? I've notice occassional occurance of the same messages, and they seem to be comming from only certain type of ATA's. I'm suspecting it's ATA related, but I don't have enough evidence to prove so yet. Andy On 3/14/06, Dan Morin [EMAIL PROTECTED] wrote: The past two

Re: [Asterisk-Users] Asterisk and T38 Fax

Hi, I tried to connect two T.38 capable SIP ATA's through Asterisk. I had canreinvite=yes and the 2 ATA's did talk directly to each other, but fax still failed. From Ethereal captures, I think the problem was when the originating ATA is ready to set up a T.38 session, it sends a message to the

Re: [Asterisk-Users] RE: ZAP extension, DTMF?

Hi Dan, How is your echo can the issue? Did you disable the echo can and solve the DTMF issue? If you did, did it trade the DTMF issue with echo problem? It would nice if you can share your experience. Thanks. Andy On 2/14/06, Dan Elder [EMAIL PROTECTED] wrote: Please ignore my last query

Re: [Asterisk-Users] lists problem, Gmail????????

Hi, I'm using gmail, and I've been getting messages from the list and other people. Not sure what is the actual cause, but it looks to me like a subscription problem. Andy On 2/8/06, C F [EMAIL PROTECTED] wrote: Am I the only one having trouble with this list? Since the begining of the week I

Re: [Asterisk-Users] Anyway to do this?

Hi, Sorry to ask a slightly off topic question here, but I've been stuck on this for a while. My SIP ATA's are displaying callerID without problems. The problem is when a 2nd call comes in during a conversation, callwaiting callerID dosen't show up. I can only hear the callwaiting alert tones,

Re: [Asterisk-Users] How to view Q.931 Disconnect code

I replied the following and got a returned to sender message from the MAILER-DAEMON. Not sure if you got it. Here it is again... On 1/23/06, Andy Kuo [EMAIL PROTECTED] wrote: Hi, Try exten = h,1,NoOp(${HANGUPCAUSE}) in your extensions.conf Cheers. Andy On 1/23/06, Angelito Manansala

Re: [Asterisk-Users] How to view Q.931 Disconnect code

Hi, Try exten = h,1,NoOp(${HANGUPCAUSE}) in your extensions.conf Cheers. Andy On 1/23/06, Angelito Manansala [EMAIL PROTECTED] wrote: Hi there, Can anyone know how to view asterisk disconnect code.? -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +63 917 542 5807

[Asterisk-Users] Call Waiting CallerID

Hi, According to the wiki, we need to have both callwaiting=yes and callwaitingcallerid=yes , and that's what I have in zapata.conf. I can hear the call waiting alert tone when a 2nd call comes in during an established call, and I can switch between the calls without problems. However, CallerID

[Asterisk-Users] Can TE406P provide PRI to other VoIP gateways?

Hi, Does anyone know if it's possible to have 1 port of TE406P to provide T1 PRI to other VoIP gateways? I am trying to provide T1 PRI to one of our old Clarent gateways to integrate it to Asterisk. Can anyone help? Thank you. Andy ___ --Bandwidth

[Asterisk-Users] Call Waiting CallerID not showing up

Hi All, According to the wiki, we need to have both callwaiting=yes and callwaitingcallerid=yes , and that's what I have in zapata.conf. I can hear the call waiting alert tone when a 2nd call comes in during an established call, and I can switch between the calls without problems. However,

Re: [Asterisk-Users] SpanDSP not sending to fax extension.

Hi, Are you using exten = fax,1,rxfax(. in extensions.conf and faxdetect=both in zapata.conf? If yes, have you tried assigning an extension number for receiving fax? (instead of the fax extension) Andy On 1/18/06, Ken D'Ambrosio [EMAIL PROTECTED] wrote: Hi, all. I've got a fax

Re: [Asterisk-Users] Call Forwarding with Account Code.. can it be done?

I use SetAccount(${EXTEN}) when the extension gets the call. The original dialed extension will be recorded as AccountCode in CDR, before the call is forwarded. The 1st field in CDR will be the extension your customer, the 2nd will be the caller (source), the 3rd will be the forwared number. It

Re: [Asterisk-Users] Context confict question??

Hi, The one in [big-business] has higher priority than the one in [small-business]Included context has lower priority. Hope this helps. Andy On 12/2/05, Chuck Bunn [EMAIL PROTECTED] wrote: Hi,If I have an extension in a context and I have another context with thesame extension and I include the

[Asterisk-Users]can I have T1 and E1 on the same TE406 card?

Hi, We have aDigiumTE406P connected to 1 T1/PRI now. Can I put in an E1 to one of the unused ports on the same card? Thanks. Andy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users]can I have T1 and E1 on the same TE406 card?

Thank you Steve / Kevin. I'll look for the jumper on the card when I go to our co-lo. Andy On 11/25/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: Andy Kuo wrote: We have a Digium TE406P connected to 1 T1/PRI now. Can I put in an E1 to one of the unused ports on the same card? Yes, just change

Re: [Asterisk-Users] Best Communications Line for VoIP

If you need to make calls in and out to the PSTN, you need a T1/PRI. Unless you send the calls to other VoIP provider. Andy On 11/22/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: We are putting in an Asterisk VoIP solution and was wondering what the bestcommunications medium would be for this

Re: [Asterisk-Users] Asterisk to Fax Server

What Fax server are you using? On 11/21/05, Arcady Litmanovich [EMAIL PROTECTED] wrote: Hi I'm looking for following solution: Asterisk is connected to PSTN by Digium or some another card which has Fax Detection If incoming call is a fax I woud like to transfer it to External Fax server by SIP

[Asterisk-Users] GotoIf always goes to true?

Hi all, I recently found GotoIf not working right in my extensions.conf, so I write a simple test and test it on my newly installed v1.2 box. However, in all cases, GotoIf seems to always result in true. This happens to me in both ABE and V1.2 my extensions.conf : [globals]Music=123 [default]

Re: [Asterisk-Users] receive fax with asterisk

Can you tell me how you send/receive fax? ie. Fax conneted to ATA - Asterisk - Fax on PSTN? or Fax connected to Digium card - Asterisk -- Fax on PSTN? or ??? Thanks. AK On 11/16/05, Ben Higley [EMAIL PROTECTED] wrote: I have downloaded the iaxmodem package, and incorporated it

Re: [Asterisk-Users] Outgoing sound very low

have you played around with rxgain and txgain in zapata.conf? AK On 11/16/05, Abdock [EMAIL PROTECTED] wrote: Hello,I have setup asterisk with Digium TDM04B, with FXO ports. I dial in using a local line and the asterisk connects me to the SIP phone network. I can hear them loudly, but when i talk

Re: [Asterisk-Users] How to check how many G729 codec licenseinstalled

I think you can specify the priority in sip.conf in the [general] section disallow=all allow=g729 allow=ulaw allow=alaw . . . During call setup, Asterisk will negotiate for g729 first, if it's not available on the other end, it'll try ulaw next, then alaw... I'm not sure what happens when your

Re: [Asterisk-Users] Fail over?

in extensions.conf exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) exten = _X.,2,Dial(SIP/[EMAIL PROTECTED]) On 11/11/05, John E. Elkin [EMAIL PROTECTED] wrote: Maybe its already been posted, but i cant find it... I have an asterisk box running agilevoice (Customer signup and provisioning system)

Re: [Asterisk-Users] Asterisk Fax support using T.38

Lilantha I've been looking for fax solutions with Asterisk too. Unfortunately, it seems like there's no T.38support for Asterisk so far. In fact, I think there's only fax-to-email solution for * now. I'm gettingsome SIP ATA's with T.38 support next week, but I am not sure if I can somehow get

Re: [Asterisk-Users] Sensing fax with txfax

Hi, I think you should change your codec on both legs to ulaw/alaw. I've been trying to get txfax going myself too. I can receive fax with Spandsp 0.0.2pre19 on Asterisk 1.0.9 with rxfax, but whenI tried sending the received fax fileto a fax machine, I either get line error or just a blank

Re: [Asterisk-Users] asterisks talking to asterisks

I do that through SIP. Assuming your TX extensions are 10XX, and NJ extensons are 20XX On your NJ box... sip.conf [gwtx] type=friend secret=x host=10.11.12.13(your TX IP) extensions.conf [toTX] exten = _10XX,1,Dial(SIP/[EMAIL PROTECTED]) On your TX box sip.conf [gwnj] type=friend

[Asterisk-Users]how to send fax using Spandsp

Hi, I've been trying to get fax going for the last few days. I can receive fax with Spandsp 0.0.2pre19 on Asterisk 1.0.9 with rxfax, but whenI tried sending the received fax fileto a fax machine, I either get line error or just a blank page. Is anyone using Spandsp to send fax to fax machines

[Asterisk-Users] Fax between Asterisk SIP clients

Hi all, I'm looking for a fax solution with Asterisk. I would like the users to be able to hook up regular fax machines to their SIP ATA's and send/receive fax from PSTN and/or other SIP clients. My goal is: fax machines - SIP ATA - Asterisk - T1(TE406E) - fax on PSTN It looks

Re: [Asterisk-Users] OS for ABE

Weuse Fedora 3 and ABE-A.1 The pair has been workinggreat for usso far. AK On 11/2/05, Eric Alexander [EMAIL PROTECTED] wrote: We are setting up ABE for a client of ours. This is not our first Asterisk install, far from it, but it is our first time using ABE. Here is the problem, ABE only

[Asterisk-Users] spandsp patch

Hi all, I'm trying to install spandsp. I followed the instructions on http://www.soft-switch.org/installing-spandsp.html, and when I applied the patch, I got the following errors: [EMAIL PROTECTED] apps]# patch apps_makefile.patchpatching file MakefileHunk #1 FAILED at 55.Hunk #2 FAILED at

Re: [Asterisk-Users] problem with receiving faxes over cisco as5300

Hi, Sorry this does not answer your question. As I am trying to implement fax on Asterisk, can you please tell me if you are using spandsp? Are you sending fax from SIP ATA's? Thank you. AK On 10/27/05, Florian Meister [EMAIL PROTECTED] wrote: Hi,does anybody have a working sample configuration

Re: [Asterisk-Users] cdr_odbc with tds

What database server are you using? If you are using MSSQL, just use freetds without unixODBC. AK On 10/20/05, Ben merrills [EMAIL PROTECTED] wrote: Does anyone know why, using latest cvs head, freetds 0.62.1-0 andunixODBC, when running cdr_odbc, it says it's logged the call successfully,

Re: [Asterisk-Users] sip accounts

Hi, Try add [1234] ... host=dynamic or host=xxx.xxx.xxx.xxx (the client's IP) ... ... AK On 10/14/05, Kong [EMAIL PROTECTED] wrote: hi, i facing a problem here. in my sip.conf, i specify a account like this,[1234]type=friendcontext=from-sip

Re: [Asterisk-Users] asterisk log

Hi Andrea, How do you start a weekly-rotation log? Do we need to do it manually through CLI? or can we set it somewhere? Thanks. AK On 10/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thank you very muchI decided not lo lower the log information (leaving all: full

[Asterisk-Users] call waiting not working on PAP2

Hi, I have callwaiting=yes in my zapata.conf, and Call Waiting Serv: Yes in the PAP2s. However,there's sitllno callwaitingon the PAP2s. Everything else work fine. Any ideas? Am I missing something somewhere? Thank you. AK ___ --Bandwidth and

Re: [Asterisk-Users] call waiting not working on PAP2

if they are using the same terminology buts it the same hardware. On 10/13/05, Andy Kuo [EMAIL PROTECTED] wrote: Hi, I have callwaiting=yes in my zapata.conf, and Call Waiting Serv: Yes in the PAP2s. However,there's sitllno callwaitingon the PAP2s. Everything else work fine. Any ideas? Am I missing

Re: [Asterisk-Users] Modifying cmd VoicemailMain

Hi, Maybe you can record the sound file vm-five.gsm as five hour in Japanese, instead of just five. AK On 10/12/05, Kuniyoshi Murata [EMAIL PROTECTED] wrote: Dear Asterisk Users,I'm a Japanese and now configuring Voicemail.Now I need to modify the way cmd VoicemailMain works to fix language

Re: [Asterisk-Users] Noise using TE410P Rhino Channel Bank

I'm not sure where the noise is coming from, but you can change the timing source in zaptel.conf in zaptel.conf: span=1,0,0,esf,b8zs --- Asterisk is using external timing source span=1,1,0,esf,b8zs -Asterisk is providing timing to the channel bank AK On 10/7/05, Eddie [EMAIL

[Asterisk-Users] Codec issue? Dropping incompatible voice frame ...

Hi, When I call forward on PAP2, the incoming call will right the forwarded number. However, there is one-way voice problem. The caller can hear the destination(the forwarded number), but after the called party answers, the caller can't hear anything. Then the CLI produce continuous errors as

[Asterisk-Users] how do I know what codec is being used

Hi, This may be a stupid/easy question for many of you. Q. how do I know what codec is being used for a particular call or call leg? Thanks. AK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list

Re: [Asterisk-Users] success story: TE406P (quadspan with hardware echocan)

Hi Andrew, I'm using a TE406P too, and I have echocancel=yes in zapata.conf. Is this redundant? Should I take the line out? Please advice. Thanks. AK On 10/3/05, Rod Bacon [EMAIL PROTECTED] wrote: Which version of asterisk and zaptel are you using?Will they work with 1.0.9

Re: [Asterisk-Users] FreeTDS 0.63

Hi, People on the list just told me that we can only use 0.62.x AK On 10/3/05, Richard Cook [EMAIL PROTECTED] wrote: Hello, Is anyone using FreeTDS version 0.63 with *? --Richard Cook[EMAIL PROTECTED]T: 705-223-2000 x2010 ___--Bandwidth and

[Asterisk-Users] CDR and TDS

Hi, I'm trying to install FreeTDS. I followed the instructions on http://www.voip-info.org/tiki-index.php?page=FreeTDS , but still can't get it to work. I serched around trying to find instructions on it, and it seems the same info (even wording) appear on all sites I found. I downloaded

[Asterisk-Users] please help on FreeTDS (writing CDR to MS-SQL or MySQL)

Hi, I'm trying to install FreeTDS. I followed the instructions on http://www.voip-info.org/tiki-index.php?page=FreeTDS, but still can't get it to work. I serched around trying to find instructions on it, and it seems the same info (even wording) appear on all sites I found. I downloaded

Re: [Asterisk-Users] Delay in dial

Hi, Try taking out Answer in your extensions.conf. You don't need to answer before dialing a SIP channel. cheers. AK On 9/28/05, yusuf [EMAIL PROTECTED] wrote: Hi all,I am using Asterisk CVS, and I am getting a huge delay in dialing SIP.This Asterisk box is taking calls from a PABX over ZAP,

Re: [Asterisk-Users] T.38 Faxing

Hi, It sounds like Asterisk should handle T.38 fax (at least at very light load). However, I am having troubles faxing through a Linksys PAP2 NA. It doesn't seem to get through at all. Everything else (voice calls) works just fine. I am using Asterisk Business Edition ( ABE-A.1). I have

Re: [Asterisk-Users] Sipura 2000 Dial Plan

5551212 1212121212 will be matched by [2-9]xxS0, and that allows only 7 digits. That's why It does not dial for a while and then it dials 555 1212 try appending a . at the end of the dial pattern, it means repeat. ie, x.allowsrepeats of x AK On 9/27/05, Michael Blood [EMAIL PROTECTED]

Re: [Asterisk-Users] ZAP ISDN losing digits

I had similar problem using Digium TE406 card. Try update the driver. I worked for me. Good luck. AK On 9/23/05, maka [EMAIL PROTECTED] wrote: Hi all, I got into a strange problem here. I've got an asterisk box with bristuff-0.2.0-RC7k, and a HFC PCI ISDN card, running in NT mode. The ISDN card

[Asterisk-Users] problems with sending fax from SIP channels

Hi All, I'm having problem sending fax fromSIP extensions (Linksys PAP2) through Asterisk Zap channels (ISDN PRI). The SIP extensions can receive fax without problems, but sending fax fails most of the time. Does anyone have this problem? Please advice. Thank you. AK