Re: [asterisk-users] (no subject)
vi /etc/asterisk/extensions.conf On Fri, Feb 22, 2008 at 12:08 AM, sandeep [EMAIL PROTECTED] wrote: hi, how to write a advanced dial plan for example: dial to a extension(123).if the user didnot pick the call, caller should get a ivr script(Enter 1 to to dial operator and 2 to go to voicemail) If caller press 1 it should dial to the operator,else if he dials 2 it should go to the voicemail of calle's extension. thanks sandeep. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
hi, how to write a advanced dial plan for example: dial to a extension(123).if the user didnot pick the call, caller should get a ivr script(Enter 1 to to dial operator and 2 to go to voicemail) If caller press 1 it should dial to the operator,else if he dials 2 it should go to the voicemail of calle's extension. thanks sandeep.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized information regarding asterisk is coming. I am putting my h323.conf and ooh323.conf h323.conf ; The NuFone Network's ; Open H.323 driver configuration ; listenAddress=10.142.17.68 listenPort=1720 connectPort=1720 ;TCP tcpStart=1 tcpEnd=2 ;UDP udpStart=1 udpEnd=2 [general] port = 1720 bindaddr = 0.0.0.0 ; this SHALL contain a single, valid IP address for this machine ;tos=lowdelay ; ; You may specify a global default AMA flag for iaxtel calls. It must be ; one of 'default', 'omit', 'billing', or 'documentation'. These flags ; are used in the generation of call detail records. ; ;amaflags = default ; ; You may specify a default account for Call Detail Records in addition ; to specifying on a per-user basis ; ;accountcode=lss0101 ; ; You can fine tune codecs here using allow and disallow clauses ; with specific codecs. Use all to represent all formats. ; ;disallow=all ;allow=all ; turns on all installed codecs ;disallow=g723.1; Hm... Proprietary, don't use it... ;allow=gsm ; Always allow GSM, it's cool :) ;allow=ulaw ; see doc/rtp-packetization for framing options ; ; User-Input Mode (DTMF) ; ; valid entries are: rfc2833, inband ; default is rfc2833 ;dtmfmode=rfc2833 ; ; Default RTP Payload to send RFC2833 DTMF on. This is used to ; interoperate with broken gateways which cannot successfully ; negotiate a RFC2833 payload type in the TerminalCapabilitySet. ; ; You may also specify on either a per-peer or per-user basis below. ;dtmfcodec=101 ; ; Set the gatekeeper ; DISCOVER - Find the Gk address using multicast ; DISABLE - Disable the use of a GK ; IP address or Host name - The acutal IP address or hostname of your GK gatekeeper = DISABLE ;gatekeeper=10.142.17.68 ; ; ; Tell As terisk whether or not to accept Gatekeeper ; routed calls or not. Normally this should always ; be set to yes, unless you want to have finer control ; over which users are allowed access to Asterisk. ; Default: YES ; ;AllowGKRouted = yes ; ; When the channel works without gatekeeper, there is possible to ; reject calls from anonymous (not listed in users) callers. ; Default is to allow anonymous calls. ; ;AcceptAnonymous = yes ; ; Optionally you can determine a user by Source IP versus its H.323 alias. ; Default behavour is to determine user by H.323 alias. ; ;UserByAlias=no ; ; Default context gets used in siutations where you are using ; the GK routed model or no type=user was found. This gives you ; the ability to either play an invalid message or to simply not ; use user authentication at all. ; ;context=default ; ; Use this option to help Cisco (or other) gateways to setup backward voice ; path to pass inband tones to calling user (see, for example, ; http://www.cisco.com/warp/public/788/voip/ringback.html https://webmail.wipro.com/exchweb/bin/redir.asp?URL=http://www.cisco.com/warp/public/788/voip/ringback.html ) ; ; Add PROGRESS information element to SETUP message sent on outbound calls ; to notify about required backward voice path. Valid values are: ; 0 - don't add PROGRESS information element (default); ; 1 - call is not end-end ISDN, further call progress information can ;possibly be available in-band; ; 3 - origination address is non-ISDN (Cisco accepts this value only); ; 8 - in-band information or an appropriate pattern is now available; ;progress_setup = 3 ; ; Add PROGRESS information element (IE) to ALERT message sent on incoming ; calls to notify about required backwared voice path. Valid values are: ; 0 - don't add PROGRESS IE ( default); ; 8 - in-band information or an appropriate pattern is now available; ;progress_alert = 8 ; ; Generate PROGRESS message when H.323 audio path has established to create ; backward audio path at other end of a call. ;progress_audio = yes ; ; Specify how to inject non-standard information into H.323 messages. When ; the channel receives messages with tunneled information, it automatically ; enables the same option for all further outgoing messages independedly on ; options has been set by the configuration. This behavior is required, for ; example, for Cisco CallManager when Q.SIG tunneling is enabled for a ; gateway where Asterisk lives. ; The option can be used multiple times, one option per line. ;tunneling=none ; Totally disable tunneling (default) ;tunneling=cisco; ; Enable Cisco-specific tunneling ;tunneling=qsig ; Enable tunneling via Q.SIG messages ; ;-- JITTER BUFFER
[asterisk-users] (no subject)
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Re: [asterisk-users] (no subject)
Check your extensions.conf On Jan 1, 2008 11:33 AM, lists65 [EMAIL PROTECTED] wrote: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Andrew Joakimsen wrote: Check your extensions.conf Hahahahaha! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
[EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, October 31, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
What is the issue with the Grandstream? We are getting tired of Cisco issues, so we have started looking at Grandstream and they seem to be pretty good. The Polycom work well, but they seem to die after about a year or so. We bought 20 of them about 2 years ago and 7 of them have died or had buttons stop working so we had to replace them. I haven't had a single Cisco do that and we have probably 100 of them. Jim Houser wrote: We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, October 31, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
We have used the Grandstream GPX2000, HT503 and GXW4104 gateways. Quality is in all cases are on the lower end. The quality I refer to is buggy software and poor call quality. I have been involved with Telecom since the early 80s and dealt with a lot of phone systems. The Grandstream phones just plain feel cheap. Real Walmart quality, not professional business class equipment. The phone functioned ok and was super easy to setup but complaints of echo and poor volume levels were common. They may be better as we have not used them in over 6 months. We have recently used their gateways due to good pricing and their economics fit our solution base well but ran into issues with them. I believe their gateways will get improved as both are new and on early firmware releases. However, we got upset with poor support. Either no call back at all or a useless email a day later with little to no information to help solve our issue. In Grandstream's defense it may be we are just too small to matter and that's ok. We prefer to go elsewhere and deliver product that when the average user picks it up to talk on it they say this is quality stuff. Asterisk is as talented as the firm that programs it BUT the phone is crucial in the end user's system satisfaction. Regardless of what you put in the back room the phone IS the device that sets the impression to your client if you are delivering a quality solution. We would do Cisco because it is high quality but we don't care to fight with the configuration or licensing issues. We would do Polycom, and probably will, but have not had the time to jump to through the hoops needed to acquire good enough pricing to make money selling them. We feel Aastra is a good compromise in delivering quality product to make the customer happy with their decision while still making us to make some sort of small profit for our time. It's solid and provides a quality feel and function. This said, Grandstream is not junk and this is not meant to be a Grandstream rant. I would like to apologize if I jumped in too quick sounding that way. Grandstream is just the lower end of quality and should be deployed in applications where the client is willing to accept that. That's not our marketplace. If you want easy to configure, low cost, slam dunk Asterisk deployments then Grandstream works. But the end result will not be as good if you build a system with Cisco, Polycom, Snom, or Aastra. We've even tested Avaya 46XX phones on Asterisk. They sound GREAT! Probably one of the best. We just can't get Asterisk to light the messaging waiting light on the phone. Arrggg! You need to decide what your marketplace offering is and what your clients are willing to accept. If call quality is the most important then our testing shows nobody beats Polycom or Avaya. Someday we are going to beat the Avaya message waiting light issue. If quality of deskset feel is the most important factor them Avaya and Cisco stand out as best. We will not put configuration into a factor simply because the customer uses the tool we are expected to configure it to their needs. We won't sell them any device based on it being easier for us to configure. I would like to hear what people say about Snom as their sets look very nice. Sorry for the novel, all I really wanted to express is Grandstream is cheap, look before you jump. Good luck on your decision... Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Wednesday, October 31, 2007 11:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) What is the issue with the Grandstream? We are getting tired of Cisco issues, so we have started looking at Grandstream and they seem to be pretty good. The Polycom work well, but they seem to die after about a year or so. We bought 20 of them about 2 years ago and 7 of them have died or had buttons stop working so we had to replace them. I haven't had a single Cisco do that and we have probably 100 of them. Jim Houser wrote: We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard
Re: [asterisk-users] (no subject)
Honestly, Its my opinion that the Aastra phones are very lacking in the firmware department. If they could get that sorted out I wouldn't mind using them. But for now there are too many NAT issues mostly caused because they use an OLD version of Broadcom CallCtrl. Why they use an ancient version is beyond me but the phones dont even have a NAT keepalive option. They promise updates to their firmware but then they only fix minor bugs. Grandstream are ok. But as others have said their support is very lacking. I've had products of theirs behave very oddly like operate and refuse to apply any settings no matter what and not allow a factory reset... paperweight. I'd personally use Polycom in the situations where there's no NAT and the Linksys SPA-phones where you do have NAT. On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
We have Cisco 9760 for executives and Aastra 9112i for everybody else. We started with Grandstream, don't remember the model, cost around $80 USD but it had bad audio quality and echo problems (running asterisk 1.09). The quality of construction felt poor, like a toy phone. We replaced them with the Aastra for double the cost and the quality improved dramatically. Audio quality was much better and echo problems all but eliminated. This phone also feels more solid. There are a few areas that are not perfect; the speaker phone is good not excellent and we have had to replace a couple of phones because they have stopped working. Over all I would say not bad for the price especially if they are for general use. We had to upgrade from the Aastra phones for our executives because they needed very good audio for both handset and speaker phone. We are using Cisco 9760's for them and have had no problems with quality. Plus they have a very solid feel. My question to the list is: As I need to add phones I am considering buying used Cisco 9760's. Is there any difference with the 9760G? I have heard that the 9761's have even better audio quality. Our main requirement is audio quality, our users do not need a lot of features on their phones. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Wednesday, October 31, 2007 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) What is the issue with the Grandstream? We are getting tired of Cisco issues, so we have started looking at Grandstream and they seem to be pretty good. The Polycom work well, but they seem to die after about a year or so. We bought 20 of them about 2 years ago and 7 of them have died or had buttons stop working so we had to replace them. I haven't had a single Cisco do that and we have probably 100 of them. Jim Houser wrote: We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, October 31, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
What is the use case? Linksys, Polycom, Snom, and Aastra all have their strengths and weaknesses. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, October 29, 2007 10:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Stay away from Cisco they just don't work for the price, if it would be in the price range of a Grandstream phone I would tell you go for it, but at the current price its just not worth it. Aastra, Polycom or linksys all work for me. Never tried Snom before. On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
I've had experience with Linksys and Polycom. Either one is easy enough to provision. Took me a while to understand how to provision Polycom. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 30 October 2007 3:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hello, I am looking for an Asterisk consultant for occasional support on an asterisk phone system located in San Francisco. It would probably be primary remote support, but we may need some on site support occasionally. Please let me know if you are interested and available. Thanks, Niki Selken Junior Systems Administrator Colorful Expressions [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Motherboard with SATA RAID1 support That's a mulit-port SATA controller with RAID in the driver (software). 256 MB RAM Use a little more RAM. digium PRI/E1 card Is there any reason you aren't using Sangoma cards? 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) None. 2. If I use Hardware based RAID 1, what would be the impact to the system? A PCI slot. 3. According to your practical experiance what is the ideal solution among both options? Software RAID works fine. -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] better subject needed [was: Re: Query1]
On Sat, Jul 28, 2007 at 12:03:33PM +0530, [EMAIL PROTECTED] wrote: Hi, I am facing problem in configuring D-channel for TE120P card.I did the following things /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf [ snip ] /etc/asterisk/zaptel.conf . Hmm.. sounds familiar. Haven't I answered it already. I also recall someone replying to it just today... You have already posted that question. Two of us have already posteated follow-ups on it. Please reply to (at least one) of them rather than re-posting your question. Furthermore, your posts have no meaningful subject. This post could use a subject such as: problem in configuring D-channel for TE120P card Or even just: configuring D-channel for TE120P card -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi,
I am facing some issues while using MixMonitor and StopMonitor. My
extensions logic is attached below:
exten = s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b)
exten = s,2,Dial(SIP/101,13)
exten = s,3,StopMonitor()
exten = s,4,NoOp(Dial Status: ${DIALSTATUS})
exten = s,5,Goto(sss-${DIALSTATUS},1)
exten = sss-NOANSWER,1,VoiceMail([EMAIL PROTECTED])
exten = sss-NOANSWER,2,Goto(salesivr,s,4)
As evident from the dialplan I only want to record the call when
Dial(SIP/101,13) is successful.
After that I disable recording by issuing the StopMonitor command. Now
the problem is that when the status of dial is NOANSWER the voicemail
recording is also recorded and saved.
It is only after I hangup that I see the following print on the console
End MixMonitor Recording SIP/192.168.0.10.172-081c67c0
I want monitor to be disabled on priority s,3. Can someone please
point out what I am doing wrong here.
Regards,
Asif
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RE: [asterisk-users] (no subject)
Hi Guy,. you should at least put a subject any way follow this link http://nerdvittles.com/index.php?p=134 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Mon, 11 Jun 2007 18:36:54 +0530 Subject: [asterisk-users] (no subject) Hi, please help me in developing and reading Text through IVR application using asterisk. can any one help me at highlevel on this, other than using SPANDSP application. Regards K.Rajesh. _ Tried the new MSN Messenger? It’s cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in ___ --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 _ With Windows Live Hotmail, you can personalize your inbox with your favorite color. www.windowslive-hotmail.com/learnmore/personalize.html?locale=en-usocid=TXT_TAGLM_HMWL_reten_addcolor_0607___ --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)
Hi, please help me in developing and reading Text through IVR application using asterisk. can any one help me at highlevel on this, other than using SPANDSP application. Regards K.Rajesh. _ Tried the new MSN Messenger? Its cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in ___ --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] (no subject)
That made all the difference! Thanks again! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Wednesday, May 30, 2007 6:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] (no subject) Thanks; I have made the change and I will try it tomorrow! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cristian N. Bradiceanu Sent: Wednesday, May 30, 2007 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) Hi, Please take a look at http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+g otchas iax.conf The new threading model is great, but the default of 10 threads is way too low. Symptoms include total loss of audio until the channel is hung up. in general section, add: iaxthreadcount = 200 in general section, add: iaxmaxthreadcount = 1000 Hope this helps. Regards, Cristian ___ --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)
Need some help with IAX trunking.
I've got six systems:
AsteriskM (main)
___|
| || | |
Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5
AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other Asterisk
boxes are using ztdummy for timing, they are all using IAX trunking.
My calls come in over Sip or Zap to asteriskm and are routed to one of the
asteriskN servers based on load. The routing is done by a small AGI script
that gets the current load from a monitoring machine and then changes the
priority. Dialplan snippet:
--- Snippet ---
exten = _X.,1,AGI(manager.agi)
exten = _X.,100,Dial(IAX2/asterisk1/${EXTEN})
exten = _X.,200,Dial(IAX2/asterisk2/${EXTEN})
exten = _X.,300,Dial(IAX2/asterisk3/${EXTEN})
exten = _X.,400,Dial(IAX2/asterisk4/${EXTEN})
exten = _X.,500,Dial(IAX2/asterisk5/${EXTEN})
--- Snippet ---
This works fine for a few calls. I'm using the SIPp package to generate a
10-25 simultaneous call load. Every once in a while I starting seeing loads
of error messages on AsteriskM's console:
chan_iax2.c:7054 socket_process: Peer 'asterisk4' is now UNREACHABLE! Time:
2
chan_iax2.c:6216 socket_read: Out of idle IAX2 threads for I/O, pausing!
chan_iax2.c:909 __schedule_action: Out of idle IAX2 threads for scheduling!
chan_iax2.c:7054 socket_process: Peer 'asterisk4' is now REACHABLE! Time:
134
chan_iax2.c:6216 socket_read: Out of idle IAX2 threads for I/O, pausing!
That is just a small example, I may have 50-100 of these type of messages
scroll very quickly. If I give the system a minute everything goes back to
normal.
I would like some one who is very knowledgeable about IAX to assist me with
this problem. If someone knows a lot about IAX optimization and is willing
to work with me I would be willing to pay for their time.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
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Re: [asterisk-users] (no subject)
Hi,
Please take a look at
http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+gotchas
iax.conf The new threading model is great, but the default of 10 threads is
way too low. Symptoms include total loss of audio until the channel is hung
up.
- in general section, add: iaxthreadcount = 200
- in general section, add: iaxmaxthreadcount = 1000
Hope this helps.
Regards,
Cristian
On 5/30/07, David Ruggles [EMAIL PROTECTED] wrote:
Need some help with IAX trunking.
I've got six systems:
AsteriskM (main)
___|
| || | |
Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5
AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other Asterisk
boxes are using ztdummy for timing, they are all using IAX trunking.
My calls come in over Sip or Zap to asteriskm and are routed to one of the
asteriskN servers based on load. The routing is done by a small AGI script
that gets the current load from a monitoring machine and then changes the
priority. Dialplan snippet:
--- Snippet ---
exten = _X.,1,AGI(manager.agi)
exten = _X.,100,Dial(IAX2/asterisk1/${EXTEN})
exten = _X.,200,Dial(IAX2/asterisk2/${EXTEN})
exten = _X.,300,Dial(IAX2/asterisk3/${EXTEN})
exten = _X.,400,Dial(IAX2/asterisk4/${EXTEN})
exten = _X.,500,Dial(IAX2/asterisk5/${EXTEN})
--- Snippet ---
This works fine for a few calls. I'm using the SIPp package to generate a
10-25 simultaneous call load. Every once in a while I starting seeing
loads
of error messages on AsteriskM's console:
chan_iax2.c:7054 socket_process: Peer 'asterisk4' is now UNREACHABLE!
Time:
2
chan_iax2.c:6216 socket_read: Out of idle IAX2 threads for I/O, pausing!
chan_iax2.c:909 __schedule_action: Out of idle IAX2 threads for
scheduling!
chan_iax2.c:7054 socket_process: Peer 'asterisk4' is now REACHABLE! Time:
134
chan_iax2.c:6216 socket_read: Out of idle IAX2 threads for I/O, pausing!
That is just a small example, I may have 50-100 of these type of messages
scroll very quickly. If I give the system a minute everything goes back to
normal.
I would like some one who is very knowledgeable about IAX to assist me
with
this problem. If someone knows a lot about IAX optimization and is willing
to work with me I would be willing to pay for their time.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
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RE: [asterisk-users] (no subject)
Thanks; I have made the change and I will try it tomorrow! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cristian N. Bradiceanu Sent: Wednesday, May 30, 2007 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) Hi, Please take a look at http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+g otchas iax.conf The new threading model is great, but the default of 10 threads is way too low. Symptoms include total loss of audio until the channel is hung up. * in general section, add: iaxthreadcount = 200 * in general section, add: iaxmaxthreadcount = 1000 Hope this helps. Regards, Cristian ___ --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)
Hello, Did someone have a solution for a line fax detection for outgoing call For exemple I call number 0123456789 - if it is a fax then redirect to extension A - if it is a line then redirect to exention B whats ia want its somthing like AMD application that i use for the answering machine . http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD search in the wiki give this application : http://www.voip-info.org/wiki/view/NVFaxDetect Did somene use it ? any feed back ? Sorry for the English and thanks for your help ___ --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)
Hello , iam having 6 asterik cards on three different servers I am using asterisk 1.2.14 with zaptel 1.2.12 on fedora core 5 (2.6.17.1). now every 3 days i need to rmmod/modprobe wctdm driver to detect the call. callers wont get the IVR prompt. after rmmod and modprobe the wctdm ..it works fine. earlier i wanted to restart every day...so i removed the CLI detection on British telecom line.now its happening evry 3 days..i have used this part in additon to normal config. but it gave that error..everyday..(asterisk didnt detect the incming call) then i remove this part...now that happeing evry 3 days..(ima connected to British telecom PSTN). i have enabled loadzone=uk.. usecallerid=yes cidsignalling=v23 cidstart=polarity this is my zaptel config.. (NO CLI detection enabled) signalling=fxs_ks busydetect=yes busycount=8 threewaycalling=yes group=1 context=sip echocancel=yes channel= 1-8 echocancelwhenbridged=yes echotraining=20 echotraining=yes dtmfmode=rfc2833 rxgain=4.0 txgain=4.0 My fxo cards are connected to British telecom . can it be a problem with BT singnaling..?? because asterisk verisn 1.07 worked without any erros.. or can it be a problem with the card ? many thanks, Tharanga ___ --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)
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Re: [asterisk-users] (no subject)
You seem to have misplaced your message/comment/question. ___ --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)
hi every body, i m new to this mail list, and also with asterisk IPBX, i havr digium TE110P card, can someone till me if he has an particular experience with this card, kind of bugs, problems... kind regards Younss ___ --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)
All: I am looking to move cell phone providers. I have acquired the new cell phone and LOVE my new number but want to keep the old number as well. The new provider only will allow me to use one number or the other. They will port the old number if I want, but will keep my new number if I ask them to port the old one. Where can I go to get the old number ported from my old provider (the account is still active) and have it forwarded on to my new number (for cheap)? Sincerely, Lorell Hathcock Adaptive Data Works, LLC ___ --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)
Hello everyone! I'm planning on setting up a new system shortly and can't pick the right card... We will have 2 or 3 lines coming in and 7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... I was thinking I'd use a Dell 2.0 GHz machine as the server... If anyone has suggestions as to the benifits/problems of each card choice, I'd love to hear it. thanks Todd ___ --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] (no subject)
Todd- Asterisk wrote: Hello everyone! I'm planning on setting up a new system shortly and can't pick the right card... We will have 2 or 3 lines coming in and 7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... I was thinking I'd use a Dell 2.0 GHz machine as the server... If anyone has suggestions as to the benifits/problems of each card choice, I'd love to hear it. thanks Todd In my humble opinion, X100P's are only good for one line (and barely that). They don't work as well as the TDM400s do, and having more than one X100 card in a system is an unnecessary bombardment of interrupts. For a 2-3 line setup I would strongly suggest looking at a TDM400 or the Sangoma A200. I have used both and have been happy with both. I use a TDM400 at home and have managed to remove almost all echo with the use of fxotune and adjusting the gains. I'm using a Sangoma A200 with the on-board echo canceler for a phone system at work and have been very happy with it. The only complaint of echo on this system is on an occasional incoming call and only for the first second or two. If money is tight and you are willing to tune echo out of your system by hand, use the TDM400. If you are willing to spend the cash and don't want to have to deal with constant tweaking to remove echo, get the A200d (and make sure you download the latest drivers from sangoma). -Dave ___ --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] (no subject)
At 05:23 AM 12/14/2006, you wrote: Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... When I started down this path I choose the TDM04 and have always had occasional echo issues, not bad and not often, but it annoys the wife and one of these days I'll sell the TDM04 and replace it with the A20002D so I have hardware echo cancellation. Someone else a few months back said the same thing about all the small business installations he did because he just didn't want to have complaints and the extra $300 was a small price to pay for peace of mind. All that said, I don't have the Sangoma card yet and have never seen one so I could be blowing smoke! Ira ___ --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] (no subject)
I have been using the sangoma A200 with echo cancelation and I have been real happy. - Original Message - From: Todd- Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 14, 2006 3:23 PM Subject: [asterisk-users] (no subject) Hello everyone! I'm planning on setting up a new system shortly and can't pick the right card... We will have 2 or 3 lines coming in and 7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... I was thinking I'd use a Dell 2.0 GHz machine as the server... If anyone has suggestions as to the benifits/problems of each card choice, I'd love to hear it. thanks Todd ___ --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] (no subject)
You might want to take a look at the new 4 port FXO from Grandstream I haven't had one yet to evaluate but assuming it works it is very price competative and off-loads all the analog (TDM) stuff from your PC Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada I have been using the sangoma A200 with echo cancelation and I have been real happy. - Original Message - From: Todd- Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 14, 2006 3:23 PM Subject: [asterisk-users] (no subject) Hello everyone! I'm planning on setting up a new system shortly and can't pick the right card... We will have 2 or 3 lines coming in and 7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... I was thinking I'd use a Dell 2.0 GHz machine as the server... If anyone has suggestions as to the benifits/problems of each card choice, I'd love to hear it. thanks Todd ___ --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] (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
Re: [asterisk-users] (no subject)
We have placed 2 X 4 port cards in a Dell 2950 and it worked well - even when it was recording 50% of the calls. PaulH On Fri, 2006-11-24 at 11:54 +0600, Imran M Yousuf wrote: 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 ___ --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)
Here's a question maybe someone can help me with: My extension looks like this: exten = 1006,1,MP3Player,http://audio-mp3.ibiblio.org:8000/wcpe.mp3 When I try this extension, the following output appears in the CLI: Nov 13 12:47:51 NOTICE[8422]: app_mp3.c:111 timed_read: Poll timed out/errored out with 0 I should mention that mpg123 is installed, however the server we are using for this project doesn't have an audio card. Is this a problem? Doesn't seem to be so far (everything else works great.) Cheers! Phil Jackson -- Phil Jackson CTO, Chesapeake Medical Imaging CEO/President, SecureRAD, LLC Tel: +1 443-716-0410 GSM: +1 202-841-0090 E-mail: [EMAIL PROTECTED] ___ --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)
i am sure this came up before
but all my searches are not resulting in anything usefull
trying to setup a grandstream phone
to connect to an asterisk server
now i am outside the network (home)
on my side
settings on the phone seem to be correct
id and password, astersik server ip, port
in pf.conf
# SIP (TCP)
voip_tcp = 5060
# SIP, IAX2, IAX, RTP, MGCP (UDP)
voip_udp = {5060, 4569, 5036, 20001, 2727}
---
on the server side
same thing
plus
voip_users = ip from where i am connecting
--
can't seem to find anything else that should be opened on either side
to allow connection
--
i guess, help ?
--
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Re: [asterisk-users] (no subject)
Add a subject next time.
Are you behind a firewall where the Asterisk server is located? Have
forward ports 5060 and 1 - 2 UDP to the asterisk server?
On 11/10/06, Stas Khromoy [EMAIL PROTECTED] wrote:
i am sure this came up before
but all my searches are not resulting in anything usefull
trying to setup a grandstream phone
to connect to an asterisk server
now i am outside the network (home)
on my side
settings on the phone seem to be correct
id and password, astersik server ip, port
in pf.conf
# SIP (TCP)
voip_tcp = 5060
# SIP, IAX2, IAX, RTP, MGCP (UDP)
voip_udp = {5060, 4569, 5036, 20001, 2727}
---
on the server side
same thing
plus
voip_users = ip from where i am connecting
--
can't seem to find anything else that should be opened on either side
to allow connection
--
i guess, help ?
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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[asterisk-users] (no subject)
Hi all, the lists seems to be littered with disconnect problems using various equipment (TDM 400,Linksys etc etc.) My question is very simple and could make for good solution to Asterisk users. Since * can detect various tones according to different country standards would it be possible to disconnect on the 'off-hook' warning tone? This tone is: 1400 Hz, 2060 Hz, 2450 Hz, and 2600 Hz, at a cadence of 0.1s on, 0.1s off. is it very easy to establish if this tone is present on the line simply ask the non-asterisk end to hangup and wait on the line if you hear a loud warning tone then that is the disconnect tone!. If this tone could be detected and issued as the # then * would see this as a dialled digit and force a disconnect. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada ___ --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)
Hi All I would greatly appreciate some advice or some direction as to where to go next. I have a provider passing me incoming calls via my Session Border Controller. I am able to pass them calls fine but coming in fails with a 407 Authentication Fail error. In my sip.conf I have an entry for the provider but am not asking for a user/pass so I would expect the calls to come in and then pass to the context specified in extensions.conf: [iplcr-gw] type=peer host=xx.xx.xx.xx nat=no dtmfmode=inband context=from-iplcr insecure=invite canreinvite=yes disallow=all allow=ulaw,alaw I have tried different insecure= methods but am still getting the same error. Does anyone know what else could be causing the error or suggest some other things I should try? Many Thanks Scott ___ --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] (no subject)
You might want to repost it with a subject or you miss a lot of people seeing or opening it up. -- Original message -- From: "Scott Pinhorne" [EMAIL PROTECTED] Hi All I would greatly appreciate some advice or some direction as to where to go next. I have a provider passing me incoming calls via my Session Border Controller. I am able to pass them calls fine but coming in fails with a 407 Authentication Fail error. In my sip.conf I have an entry for the provider but am not asking for a user/pass so I would expect the calls to come in and then pass to the context specified in extensions.conf: [iplcr-gw] type=peer host=xx.xx.xx.xx nat=no dtmfmode=inband context=from-iplcr insecure=invite canreinvite=yes disallow=all allow=ulaw,alaw I have tried different insecure= methods but am still getting the same error. Does anyone know what else could be causing the error or suggest some other things I should try? Many Thanks Scott ---BeginMessage--- ___ --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 ---End Message--- ___ --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)
Date: Thu, 19 Oct 2006 09:30:38 -0500 From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk hangs up on incoming analog calls after a while To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Robert La Ferla wrote: I have been experiencing a problem where after someone calls me from an analog line, the phone call is terminated after a period of time (anywhere from 15 seconds to 15 minutes) The phone that I use to answer the call is an Aastra 9133i SIP phone. There are several other SIP extensions on the network as well as a few analog extensions on a shared FXS line. When a call comes in the analog line on the FXO, * dials all the extensions (SIP and analog.) I have a Digium card with 1 FXO and 1 FXS. Do you have callprogress=yes or busydetect=yes in your /etc/asterisk/zapata.conf ? ___ --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)
I have two questions. First I am running a t400p with three fxo ports signalling fxs (inbound CO lines). I have six polycom 501's. The problem is the amount of time the call setup takes. I have done this with Mitel phones before with a t-1 and had the same problem. My customer always complains about the call setup time. Am I doing something wrong or is this how it is. It takes up to five seconds to pickup or start ringing the CO. I would be happy to supply fake ringback if anyone knows how to do that. Second Problem is SIP Polycom phone line programming, I have read many contradicting things. How should it be provisioned to allow multiple incoming calls. How many lines,calls per line and the rest of the bull, Iknow loaded question. I am using kewl start on those three lines by the 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] (no subject)
I am going to reply inline as you asked many questions I have two questions. Sure, you do!! First I am running a t400p with three fxo ports signalling fxs (inbound CO lines). I have six polycom 501's. The problem is the amount of time the call setup takes. I have done this with Mitel phones before with a t-1 and had the same problem. My customer always complains about the call setup time. Am I doing something wrong or is this how it is. It takes up to five seconds to pickup or start ringing the CO. I would be happy to supply fake ringback if anyone knows how to do that. You can add the r command to your Dial string to fake ringback However the CLI is your friend in this case. How long does it take after dialing on the polycom for the console to reflect it is dialing? If it is right away, you may have a problem with your CO lines, and how many digits it is expecting before placing the call. Second Problem is SIP Polycom phone line programming, I have read many contradicting things. How should it be provisioned to allow multiple incoming calls. How many lines,calls per line and the rest of the bull, Iknow loaded question. I am using kewl start on those three lines by the way I set 6 lines per key on my Polycoms that works for 501 and 601. Calls just keep coming IN!!! . ___ --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)
Hi, Looking for good rates for UK Landline Mobile. Plus Saudi Arabia, UAE, India Pakistan. Thank you. John mail2web - Check your email from the web at http://mail2web.com/ . ___ --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] (no subject)
[EMAIL PROTECTED] wrote: Hi, Looking for good rates for UK Landline Mobile. Plus Saudi Arabia, UAE, India Pakistan. This is a -biz question, not -users. Also, do you realize how bad it makes you look that you can't even bother to put a subject on your mail? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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)
Hi,Is there somewhere a sample configuration for asterisk as gateway (pri - isdn)Thanks Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --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)
Best RegardsSony V. Shandy ___ --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)
i have been wondering about how the useragents work since a month or two. i have tried every document possible...could not find the answer. if anyone could tell me about the useragents, how they work, what are the factors that are considered while choosing a UA, what makes a particular UA best, it would be of great help. Thanks Regards Ramya Murthy ph-no- 9845025859 ___ --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)
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[asterisk-users] (no subject)
Sent RTP packet to 293.67.65.3:43294 (type 18, seq 59050, ts 697456, len 2) Got RTP packet from 21.98.11.200:58654 (type 18, seq 6246, ts 3559220, len 20) ANY ONE KNOWS WHAT THIS rtp DEBUD MEANS THANKS * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 I am using trixbox,I want ot disable and enable voicemail from command line At [EMAIL PROTECTED] v 2.8 I was using this command and was working successfully Database put AMPUSER/9990999 voicemail default And Database put AMPUSER.9990999 voicemail disables But at trixbox its not working Any ideas pleas Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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)
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[Asterisk-Users] (no subject)
Hey everybody, Is it alright to run two TDM400s on the same machine? If it is, how would one differentiate between the channels on each card? So, if Im running strait FXS and my first card is fxsks 1-4, would the second be fxsks 5-8? Would there be any interrupt problems? Any help would be great! Thanks! Tj ___ --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] (no subject)
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The only issues you could potentially run into is if all the modules
are FXS and they all needed to ring simultaneously... your power
supply may not be suited to handle to voltage requirements.
Sean
Ninneman, Tj wrote:
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Hey everybody,
Is it alright to run two TDM400s on the same machine? If it is,
how would one differentiate between the channels on each card? So,
if I?m running strait FXS and my first card is fxsks 1-4, would the
second be fxsks 5-8? Would there be any interrupt problems?
Any help would be great!
Thanks!
Tj
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RE: [Asterisk-Users] (no subject)
Hi Tj, yes, you can run two TDM400s (or more) on the same cpu, and the channels are 1 to 4, and 5 to 8. (also, you can set one or more groups for outgoing calls). Interrupts are the main issue. As far as possible avoids that the cards share interruptions. cheers Fabio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Ninneman, Tj Enviado el: Miercoles, 28 de Junio de 2006 12:54 p.m. Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] (no subject) Hey everybody, Is it alright to run two TDM400s on the same machine? If it is, how would one differentiate between the channels on each card? So, if I'm running strait FXS and my first card is fxsks 1-4, would the second be fxsks 5-8? Would there be any interrupt problems? Any help would be great! Thanks! Tj ___ --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] (no subject)
Sure, but if one needs that many, much better off to use the Sangoma A200 No MB problems and up to 24 channels. John Novack Fabio wrote: Hi Tj, yes, you can run two TDM400s (or more) on the same cpu, and the channels are 1 to 4, and 5 to 8. (also, you can set one or more groups for outgoing calls). Interrupts are the main issue. As far as possible avoids that the cards share interruptions. cheers Fabio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Ninneman, Tj Enviado el: Miercoles, 28 de Junio de 2006 12:54 p.m. Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] (no subject) Hey everybody, Is it alright to run two TDM400s on the same machine? If it is, how would one differentiate between the channels on each card? So, if I'm running strait FXS and my first card is fxsks 1-4, would the second be fxsks 5-8? Would there be any interrupt problems? Any help would be great! Thanks! Tj ___ --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] (no subject)
Hi,
I have the same problem with the queue configuration
When I receive 2 calls only 1 phone ring even if more agent's phone are free.
The second call will go to an other agent only if the first call is pickup.
Somebody have a solution ?This is my config file :Queue.conf[general]
;
; Global settings for call queues
;
; Persistent Members
; Store each dynamic agent in each queue in the astdb so that
; when asterisk is restarted, each agent will be automatically
; readded into their recorded queues. Default is 'yes'.
;
persistentmembers = yes
;
; Note that a timeout to fail out of a queue may be passed as part of
; an application call from extensions.conf:
; Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout])
; example: Queue(dave|t|||45)
autofill = yes
[ticketix]
;
; A sample call queue
;
; Musiconhold sets which music applies for this particular
; call queue (configure classes in musiconhold.conf)
;
autofill=yes
musiconhold = default
;
; An announcement may be specified which is played for the member as
; soon as they answer a call, typically to indicate to them which queue
; this call should be answered as, so that agents or members who are
; listening to more than one queue can differentiated how they should
; engage the customer
;
;announce = queue-ticketix
;
; A strategy may be specified. Valid strategies include:
;
; ringall - ring all available channels until one answers (default)
; roundrobin - take turns ringing each available interface
; leastrecent - ring interface which was least recently called by this queue
; fewestcalls - ring the one with fewest completed calls from this queue
; random - ring random interface
; rrmemory - round robin with memory, remember where we left off last ring pass
;
strategy = roundrobin
;
; Second settings for service level (default 0)
; Used for service level statistics (calls answered within service level time
; frame)
servicelevel = 60
;
; A context may be specified, in which if the user types a SINGLE
; digit extension while they are in the queue, they will be taken out
; of the queue and sent to that extension in this context.
;
;context = qoutcon
;
; How long do we let the phone ring before we consider this a timeout...
;
timeout = 15
;
; How long do we wait before trying all the members again?
;
retry = 5
;
; Weight of queue - when compared to other queues, higher weights get
; first shot at available channels when the same channel is included in
; more than one queue.
;
;weight=0
;
; After a successful call, how long to wait before sending a potentially
; free member another call (default is 0, or no delay)
;
wrapuptime=15
;
; Maximum number of people waiting in the queue (0 for unlimited)
;
maxlen = 0
;
;
; How often to announce queue position and/or estimated holdtime to caller (0=off)
;
announce-frequency = 90
;
;
; How often to make any periodic announcement (see periodic-announce)
;
periodic-announce-frequency=60
;
; Should we include estimated hold time in position announcements?
; Either yes, no, or only once.
; Hold time will be announced as the estimated time,
; or less than 2 minutes when appropriate.
;
announce-holdtime = yes
;
; What's the rounding time for the seconds?
; If this is non-zero, then we announce the seconds as well as the minutes
; rounded to this value.
;
announce-round-seconds = 10
;
; Use these sound files in making position/holdtime announcements. The
; defaults are as listed below -- change only if you need to.
;
queue-youarenext = queue-youarenext ; (You are now first in line.)
queue-thereare = queue-thereare; (There are)
queue-callswaiting = queue-callswaiting ; (calls waiting.)
queue-holdtime = queue-holdtime; (The current est. holdtime is)
queue-minutes = queue-minutes ; (minutes.)
queue-seconds = queue-seconds ; (seconds.)
queue-thankyou = queue-thankyou; (Thank you for your patience.)
queue-lessthan = queue-less-than; (less than)
queue-reporthold = queue-reporthold ; (Hold time)
periodic-announce = queue-periodic-announce ; (All reps busy / wait for next)
;
; Calls may be recorded using Asterisk's monitor resource
; This can be enabled from within the Queue application, starting recording
; when the call is actually picked up; thus, only successful calls are
; recorded, and you are not recording while people are listening to MOH.
; To enable monitoring, simply specify monitor-format; it will be disabled
; otherwise.
;
; You can specify the monitor filename with by calling
; Set(MONITOR_FILENAME=foo)
; Otherwise it will use MONITOR_FILENAME=${UNIQUEID}
;
monitor-format = wav49
;
; If you wish to have the two files joined together when the call ends, set this
; to yes.
;
monitor-join = yes
;
; This setting controls whether callers can join a queue with no members. There
; are three
[Asterisk-Users] (no subject)
I have a cisco VPN from router to router over a Data T-1. The ping times are consistently 32ms with random ping responses of 295ms -408ms about every 30 secs to a minute, I have jitter buffer enabled. The connection goes like this Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B and then T-1 to a PBX, the calls are internal so they are terminating on Toshiba digital phones. Loud crackling even happens from time to time when a Mitel SIP phone is connected to Asterisk B at that location over thye LAN with no layer three routing, but it is consistent on the IAX trunk. There is a lot of Data traffic, but thus should work regardless, I dont think the ping times are the issue. Jordan Novak Communications Technician ___ --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)
Good day, Hi! i've finish up setting * for my company and they are working reallly great, but i notice when i try to call to mobile phone, i can see the zap channels is bridging successfully but i hear nothing except for a long dialtone like tone, but calling to a regular pots line is working perfectly, could this be related to telco issues? or some tweaks to zapata.conf p.s. but i'm not having a problem on plugging POTS line to analog phone and calling to a mobile phone. My setup Asterisk 1.2.7.1 + zaptel 1.2.5 + libpri 1.2.2 + FC4 2.6.16 Hardware TDM400p + 2 FXO modules 2 scenarios given PSTN - DISA - * - ZAP - Mobile phone Softhpone - IAX2 - ZAP - Mobile phone both of them failed in my test environment :( Best regards, Joy___ --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] (no subject)
[EMAIL PROTECTED] could be a better start for beginners (but beware, the installation CD will format your HD without asking). http://asteriskathome.sourceforge.net/ On Tue, 2006-04-25 at 10:47 +0800, rommel malana wrote: Goodday, I'm an opensource fanatic and I have already installed asterisk in my mandriva linux. Actually, I'm also planning to install the asterisk management portal for GUI of asterisk. If anyone could help me guide in installing this. Thanks a mill for the help.. -Rommel- ___ --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] (no subject)
--- rommel malana [EMAIL PROTECTED] wrote: Goodday, I'm an opensource fanatic and I have already installed asterisk in my mandriva linux. Actually, I'm also planning to install the asterisk management portal for GUI of asterisk. If anyone could help me guide in installing this. Thanks a mill for the help.. -Rommel- Rommel, You should read the book Asterisk: The future of telephony (I believe is the name). There is a PDF of it available online. Do a google search and you should find it. Dovid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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)
Goodday, I'm an opensource fanatic and I have already installed asterisk in my mandriva linux. Actually, I'm also planning to install the asterisk management portal for GUI of asterisk. If anyone could help me guide in installing this. Thanks a mill for the help.. -Rommel- ___ --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] (no subject)
Please make sure to write a subject line. Thank You On 4/24/06, rommel malana [EMAIL PROTECTED] wrote: Goodday, I'm an opensource fanatic and I have already installed asterisk in my mandriva linux. Actually, I'm also planning to install the asterisk management portal for GUI of asterisk. If anyone could help me guide in installing this. Thanks a mill for the help.. -Rommel- ___ --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] (no subject)
Currently, compiling asterisk on an Itanium fails with the GSM codec.
All I could find on Google was a hack to basically remove GSM from the
build which is not an option for some. This patch will allow it to
compile and seems to work perfectly.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
--- Makefile2006-03-12 12:57:37.0 -0500
+++ ../../../../asterisk-1.2.6/codecs/gsm/Makefile 2006-04-12
15:11:19.0 -0400
@@ -45,6 +45,7 @@
ifneq ($(shell uname -m),ppc64)
ifneq ($(shell uname -m),alpha)
ifneq ($(shell uname -m),armv4l)
+ifneq ($(shell uname -m),ia64)
ifneq (${PROC},sparc64)
ifneq (${PROC},arm)
ifneq (${PROC},ppc)
@@ -62,6 +63,7 @@
endif
endif
endif
+endif
#The problem with sparc is the best stuff is in newer versions of gcc
(post 3.0) only.
#This works for even old (2.96) versions of gcc and provides a small
boost either way.
@@ -233,6 +235,7 @@
ifneq ($(shell uname -m),ppc)
ifneq ($(shell uname -m),ppc64)
ifneq ($(shell uname -m),alpha)
+ifneq ($(shell uname -m),ia64)
ifneq ($(shell uname -m),armv4l)
ifneq ($(shell uname -m),sparc64)
ifneq (${PROC},arm)
@@ -247,6 +250,7 @@
endif
endif
endif
+endif
TOAST_SOURCES = $(SRC)/toast.c \
$(SRC)/toast_lin.c \
@@ -297,6 +301,7 @@
ifneq ($(shell uname -m), ppc)
ifneq ($(shell uname -m), ppc64)
ifneq ($(shell uname -m), alpha)
+ifneq ($(shell uname -m), ia64)
ifneq ($(shell uname -m), sparc64)
ifneq ($(shell uname -m), armv4l)
ifneq ($(shell uname -m), parisc)
@@ -309,6 +314,7 @@
endif
endif
endif
+endif
TOAST_OBJECTS =$(SRC)/toast.o \
$(SRC)/toast_lin.o \
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[Asterisk-Users] (no subject)
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I have had the exact same problem last week. I have not yet solved it. So instead I am using ooh323, but would prefer to use oh323. Can anyone help? I'm glad that I'm not the only one :)) Hopefully we'll find solution to this problem. -- Tomislav Parcina tparcina#lama.hr ___ --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)
Hi, I'm using IPSwitchboard v 1.8.10, a sort of Operator Panel, to monitor my Asterisk's extensions. Recently I noticed that on the official site (http://ipswitchboard.thorben.dk/), where I downloaded the software some weeks ago, this project is no longer supported. Is there anyone that can say me where I can find the Italian version of IPswitchboard or if there is a way to translate the its messages? Thanks in advance, Marco. ___ --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)
Hi everybody. Yesterday I fix typo in spinlock.h and compiled zaptel. But today I have problems with soft phones. I tried to recompile zaptel and it showed errors again. So I don't understand what now it needs. Brings words and photos together (easily) with PhotoMail - it's free and works with Yahoo! Mail.___ --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)
Does anyone have a DISA alternative? I currently use the line: exten = s,16,DISA(no-password|from-internal) however that just drops a user at a dial tone, what I would like to do is prompt user for number to dial, followed by the # key, and then have asterisk dial out. Can this be accomplished by the DIAL command? Reasons for doing this: Sounds better more professional Ensure proper formatting of number ___ --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)
YUP, this is the way that asterisk works. It is going to quelch all DTMF that goes out via a SIP gateway via asterisk. I spent a long time working this through and it has to do with the way that asterisk deals with DTMF and the DSP.c module that sits inband to the RTP/audio stream. There is a flag called DSP_DIGITMODE_NOQUELCH that is broken that might allow the inband DTMF after answer to work on inband but its broke. If you do not use asterisk as your gate/ to/from the PSTN you are going to have a issue with DTMF after connect. There are a couple of kludges that can get it to work part of the time. But from my experiance DTMF is not handled correctly in asterisk if you use any gateway other then asterisk. IE: you use a cisco or TNT as your gateway to/from the PSTN via SIP and asterisk to talk to the 2500 type phones. -larry Message: 1 Date: Thu, 16 Mar 2006 12:36:45 -0800 From: Martin Joseph [EMAIL PROTECTED] Subject: [Asterisk-Users] RFC 2833 and SIP? DTMF? What am I not getting? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; format=flowed Hi again, I am trying to get my DTMF to use RFC 2833 rather then inband, so that I can utilize lower bandwidth codecs through my FXO. After much tinkering I was able to get my gateway (wellgate 3701A) configured to a point where I have some success, but no real joy. I have configured the RTP Payload type (or RFC2833 Payload type) to 101. I don't have a clue what this means, but I took the 101 from my AG168V ATA's configuration screen, as I know that device seemed to work fine through the old HT-488 fxo(via rfc2833). I then changed my asterisk extensions for both the FXS and FXO on the wellgate to include dtmfmode=rfc2833. This has brought me to a point where both my hardphones (ATA's) and my softphones (IAXcomm, or JackenIAX) work perfectly with comedian mail. To me this means that asterisk is properly getting the RFC2833 events from the user agents. BUT, if I try to dial out the FXO, none of my phones (hard or soft) produce working touchtones for a PSTN based voicemail system. Even stranger to me, is the fact that from the phone connected to the FXS on the wellgate I can hear tones(listening on a called line), but they sound kind rough at the edges. From the AG168V I hear no tones, but what seems to be blown out tones (ie overdriven volume). From the IAX softphones I hear no tones at all just clicks! Now I would have guessed that the FXO would be doing the conversion of the RFC2833 to inband, so that I thought all the tones should sound the same from any phone? Apparently this isn't the case at all. Thanks to all of you for any help understanding and or debugging this mess. Marty PS I spent a good deal of time adjusting the DTMF volume for the wellgate FXS/FXO hoping this might help before I discovered the variety of non working DTMF being generated. -- -- BEGIN:VCARD VERSION:3.0 fn:Larry Linde n:Linde;Larry org:Image Manipulation Systems Inc. adr;TYPE=dom:;;420 N 5th St. Suite #865;Minneapolis;Mn;55401 email;TYPE=internet:[EMAIL PROTECTED] tel;TYPE=work:612-746-5706 tel;TYPE=fax:612-746-5781 tel;TYPE=cell:763-438-1781 x-mozilla-html:TRUE url:http://www.imageman.com X-EVOLUTION-FILE-AS:Linde\, Larry UID:pas-id-440C50B70001 REV:2006-03-06T15:09:43Z END:VCARD ___ --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)
Hi, to all, i am new in the list and i am interest to deploy a sistem with asterisk i have a PC with a Suse Linux 8.2 and also i have Dialogic VFX card with 4 analog port. From where a can get Dialogic Driver for linux. From ware a mast beging to resolve the problem the project to implement VoIP Gateway. Savvas. _ 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
RE: [Asterisk-Users] (no subject)
Hi Im also new but you should know very well all the interfaces you are going to connect the sistem, the number of users you'll have (hardware requeriments), know a lot about the soft/hardphones you'll use and download the asterisk handbook or the big one (i don't remember the name) Good luck Jose Manuel Cortes David X Semestre Ingenieria Electronica PONTIFICIA UNIVERSIDAD JAVERIANA De: [EMAIL PROTECTED] en nombre de Savvas Gavriel Enviado el: Mié 15/03/2006 15:12 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] (no subject) Hi, to all, i am new in the list and i am interest to deploy a sistem with asterisk i have a PC with a Suse Linux 8.2 and also i have Dialogic VFX card with 4 analog port. From where a can get Dialogic Driver for linux. From ware a mast beging to resolve the problem the project to implement VoIP Gateway. Savvas. _ 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 winmail.dat___ --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] (no subject)
AFIAK, they can't - we would like to do the same thing, but it's not possible with patching the source. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 10-Mar-06, at 7:56 PM, btb wrote: can the default voicemail folders (old, work, friends, etc.) be changed? for example, i'd like to configure asterisk so that there are only folders called friends and old. thanks -ben ___ --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] (no subject)
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[Asterisk-Users] (no subject)
can the default voicemail folders (old, work, friends, etc.) be changed? for example, i'd like to configure asterisk so that there are only folders called friends and old. thanks -ben ___ --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)
HiDoes someone have a better sql query for selecting the provider used by LCDial application than the one proposed in the tgz ? It's far from working well with most of price lists.I tried to tweak it somehow with more or less success.Regards, Michel -- Michel Luczak[EMAIL PROTECTED]___ --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)
hi all, see i have problem with PC(any sip phone which registered to fwd.pulver.com) to phone(my zap where it has been registered by modifying sip.conf) my zap detects RBT but i am not able to listen to the voice,this happened when i tried with ECHO of fwd.pulver.com i dont know wat to do plz help..me if u want detail abt this i can send u bye ___ --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)
Hi, This is test mail. ___ --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)
asterisk-users@lists.digium.com ___ --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)
Mauricio, Yes it is. However I would not use analog phones. Your cheapest option would be to use softphones on a computer. If you wanted to use physical phones you have a few options. 1)Get two ATA's (device that you plug in to the LAN on your end and by your friend to the internet). This is probably the cheapest solution. You can plug in a "regular" analog phone in to the ATA device. 2)Use softphones that work on a computer 3)Get a TDM400P with one FXS port - this will cost a lot and your friend will need an ATA or VOIP phone on his end - This solution is howver worth it if you want to connect asterisk to your home line. 4)Get two VOIP phones. This sounds like the most sense. It will cost slightly more than ATA devices but they are much easier to use then POTS phones. Hope this helps and sorry if I am not to clear in the email A little tired. Regards, Dovid ___ --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)
Hi all,
I am testing my hands on asterisk , but got stuck. Let me tell you i am
only using its VOIP functionlities
I have registered the asterisk server at a remote proxy server. My clients
registered at asterisk server can make outgoing calls , but the calls made
from outside is not incoming to any extension.
I have written
user:[EMAIL PROTECTED]/1234
in sip.conf.
and 1234are defined as
[1234]
type=friend
host=dynamic
context=test_in
user=phone
regexten=1234
in extensions.conf i am using
[test_in]
exten= 1236,1,Dial(SIP/sandhu)
exten= 1235,1,Dial(SIP/1235)
exten= _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten= 1234,1,Dial(SIP/1234)
My clients are on Xlite softphone.
Can anybody help out ?/
Abhishek
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Hi to all, the following is the last thing we see from Asterisk befor it crashes: $$$ find_chan_holded: No channel found for oad:017670014533 dad:7051538 -- ch-state CONNECTED, bc-holded 0 $$$ Bchan deActivated addr 51400101 -- cause 16 I SEND:RELEASE port:1 pid:88 mode:TE addr:51400101 -- l3id:20176 cause:16 ocause:16 oad2:017670015633 dad2:7051538 channel:1 port:1 BCHAN: DeACT Conf I IND :RELEASE_COMPLETE pid:88 mode:TE addr:51400101 port:1 -- l3id:20176 cause:-1 dad:7051538 oad:017670015633 channel:1 port:1 -- cause -1 * RELEASING CHANNEL pid:88 ctx:aixtema-incoming dad:7051538 oad:017670014533 state: CONNECTED -- * State Down -- Setting AST State to down * -- In State Default == Spawn extension (aixtema-incoming, 7051538, 1) exited non-zero on 'mISDN/1/017670014533-1' * -- Queue Hangup misdn_hangup called, without chan_list obj. Ouch ... error while writing audio data: : Broken pipe These crashes happen up to five times a day. We are pretty much clueless as to what is happening here. Any help is highly appreciated :-) Rgds, Philip. ___ --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] (no subject)
See http://www.iaxtel.com/setup.html 2005/12/2, P.G.C.K. Nirukshitha [EMAIL PROTECTED]: Dear Sir I have configured two asterisk Boxes.Then I need to communicate these asterisk boxes via the IAX.It is better if you can help me to configure two boxes to communicate via asterisk. Thanks Nirukshitha Gamage -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. ___ --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 -- Giovanni Miano ___ --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] (no subject)
See http://www.iaxtel.com/setup.html 2005/12/2, Lakmal [EMAIL PROTECTED]: Hi all, I have configured two asterisk Boxes.Then I need to communicate these asterisk boxes via the IAX.It is better if you can help me to configure two boxes to communicate via asterisk Thanks, Ishanka. - Original Message - From: Branko Samardzic [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, December 02, 2005 10:43 AM Subject: [Asterisk-Users] IAX trunking frequency parameter works only oninitiator side Hi, I am experimenting with trunkfreq parameter. When it is 20ms I can see both parties in IAX session sending IAX frames every 20ms. When I modify this parameter to 40ms then I can see that only server that initiated IAX connection works properly (i.e. sends IAX frames every 40ms while other side still sends IAX frames at 20ms per frame rate). I disabled jitter buffers on both sides and I use speex codec. Here is tcp dump of IAX traffic: 23:26:45.972072 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:45.976295 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:45.996264 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.006742 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.016270 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.036254 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.047891 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.056248 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.076286 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.091255 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.096262 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.116243 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.127494 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.136242 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 SERVER_A initiates connection while SERVER_B answers. SERVER_A iax.conf file === [SERVER_B] disallow=all allow=speex jitterbuffer=no dropcount=2 maxjitterbuffer=200 maxexcessbuffer=100 minexcessbuffer=60 jittershrinkrate=1 trunkfreq=40; How frequently to send trunk msgs (in ms) context = foo secret=zYX9VUt auth=md5 type=friend host=SERVER_B_IP_ADDRESS trunk=yes SERVER_B iax.conf file === [SERVER_B] disallow=all allow=speex jitterbuffer=no dropcount=2 maxjitterbuffer=200 maxexcessbuffer=100 minexcessbuffer=60 jittershrinkrate=1 trunkfreq=40; How frequently to send trunk msgs (in ms) context = default secret=zYX9V auth=md5 type=friend host=SERVER_B_IP_ADDRESS trunk=yes Any idea as to why trunking frequency is not symmetrical? Any help is appreciated ___ --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 -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. ___ --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 -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com
[Asterisk-Users] (no subject)
Hi all, I have configured two asterisk Boxes.Then I need to communicate these asterisk boxes via the IAX.It is better if you can help me to configure two boxes to communicate via asterisk Thanks, Ishanka. - Original Message - From: Branko Samardzic [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, December 02, 2005 10:43 AM Subject: [Asterisk-Users] IAX trunking frequency parameter works only oninitiator side Hi, I am experimenting with trunkfreq parameter. When it is 20ms I can see both parties in IAX session sending IAX frames every 20ms. When I modify this parameter to 40ms then I can see that only server that initiated IAX connection works properly (i.e. sends IAX frames every 40ms while other side still sends IAX frames at 20ms per frame rate). I disabled jitter buffers on both sides and I use speex codec. Here is tcp dump of IAX traffic: 23:26:45.972072 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:45.976295 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:45.996264 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.006742 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.016270 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.036254 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.047891 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.056248 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.076286 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.091255 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.096262 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.116243 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.127494 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.136242 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 SERVER_A initiates connection while SERVER_B answers. SERVER_A iax.conf file === [SERVER_B] disallow=all allow=speex jitterbuffer=no dropcount=2 maxjitterbuffer=200 maxexcessbuffer=100 minexcessbuffer=60 jittershrinkrate=1 trunkfreq=40; How frequently to send trunk msgs (in ms) context = foo secret=zYX9VUt auth=md5 type=friend host=SERVER_B_IP_ADDRESS trunk=yes SERVER_B iax.conf file === [SERVER_B] disallow=all allow=speex jitterbuffer=no dropcount=2 maxjitterbuffer=200 maxexcessbuffer=100 minexcessbuffer=60 jittershrinkrate=1 trunkfreq=40; How frequently to send trunk msgs (in ms) context = default secret=zYX9V auth=md5 type=friend host=SERVER_B_IP_ADDRESS trunk=yes Any idea as to why trunking frequency is not symmetrical? Any help is appreciated ___ --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 -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. ___ --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 Sir I have configured two asterisk Boxes.Then I need to communicate these asterisk boxes via the IAX.It is better if you can help me to configure two boxes to communicate via asterisk. Thanks Nirukshitha Gamage -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. ___ --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)
When dialing in after hours callers get to use the directory. I know that if you put h or H with a Dial() command you get the behavior of being able to terminate a call by pressing *. However, nowhere in the entire extensions.conf does there appear the h or H option, so I know it is not that. features.conf ? So zapata.conf does define the affected Zaip channels as belonging to callgroup=1. However, they are not defined as a pickup group (that would be odd behavior). Features.conf does define the pickup extensions for pickupextension to be *8. I still don't see how that could possibly affect a caller...? Are there any Directory() gurus out there that can help? BEGIN:VCARD VERSION:2.1 N:Torrenga;Brent;August;Mr. FN:Brent August Torrenga ORG:Torrenga Engineering, Inc. TITLE:Designer TEL;WORK;VOICE:(219) 836-8918 TEL;WORK;FAX:(219) 836-1138 ADR;WORK:;;907 Ridge Road;Munster;IN;46321-1771 LABEL;WORK;ENCODING=QUOTED-PRINTABLE:907 Ridge Road=0D=0AMunster, IN 46321-1771 URL;WORK:http://www.torrenga.com EMAIL;PREF;INTERNET:[EMAIL PROTECTED] REV:20040209T215756Z END:VCARD ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] (no subject)
Hi, I seek solution for hotel management and billing solution. but I do not know which to choose between Astbill or Asterbill ? if you have council. Thx David ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] (no subject)
I have a Wildcard TDM400P card being used with Asterisk. For some reason, incoming PSTN calls are getting delayed before they ring through on the Asterisk PBX to an extension. The calling party hears an initial ring tone and then a click noise, at which point it will then actually starts to ring the target extension. I had done some research and saw similar problems that seemed to relate to caller ID so turned it off but still had the same one ring, click delay problem. I've turned it back on and verified the behavior is the same so I'm not sure this is the problem. Essentially I'd like the call to ring through immediately and not perform this one ring click until the call is routed to the correct line. Has anyone else seen this and is there a way to fix the problem? Please let me know if I can provide any additional information. -Roger ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] (no subject)
I am having the same problem, but on both PSTN and a Voicepulse Connect IAX line. PSTN rings clicks dead air, then rings and connects, IAX just clicks, has dead air, rings and connects. Don't have a clue on how to fix it though. Greg Roger Johnsen wrote: I have a Wildcard TDM400P card being used with Asterisk. For some reason, incoming PSTN calls are getting delayed before they ring through on the Asterisk PBX to an extension. The calling party hears an initial ring tone and then a click noise, at which point it will then actually starts to ring the target extension. I had done some research and saw similar problems that seemed to relate to caller ID so turned it off but still had the same one ring, click delay problem. I've turned it back on and verified the behavior is the same so I'm not sure this is the problem. Essentially I'd like the call to ring through immediately and not perform this one ring click until the call is routed to the correct line. Has anyone else seen this and is there a way to fix the problem? Please let me know if I can provide any additional information. -Roger ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] (no subject)
My Polycom IP301 hangs on Processing Cfg... Here is the boot log: 0930155446|so |4|00|-- Initial log entry -- 0930155446|so |4|00|+++ Note that bootrom log times are in GMT +++ 0930155446|wdog |4|00|Initial log entry 0930155446|cfg |4|00|Initial log entry 0930155446|copy |4|00|Initial log entry 0930155446|cdp |4|00|Initial log entry 0930155446|cdp |5|00|CDP is DISABLED. 0930155446|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 0930155446|so |3|00|Platform: Model=SoundPoint IP 301, Assembly=2345-11300-010 Rev=A 0930155446|so |3|00|Platform: Board=2345-11300-010 A 0930155446|so |3|00|Platform: MAC=0004f2022609, IP=Unknown, Subnet Mask=Unknown 0930155446|so |3|00|Platform: BootBlock=2.5.0 (11300_010) 06-Nov-04 08:07 0930155446|so |3|00|Application, main: Label=BOOT, Version=2.6.1.0003 04-Dec-04 14:38 0930155446|so |3|00|Application, main: P/N=3150-11069-261 0930155446|app1 |4|00|Initial log entry. 0930155447|so |3|00|Link status is Net up, PC down. 0930155455|app1 |3|00|Using resolver server 192.168.222.4 and domain local. 0930155455|app1 |3|00|DHCP returned result 0x287 from server 192.168.222.4. 0930155455|app1 |3|00| Phone IP address is 192.168.222.202. 0930155455|app1 |3|00| Subnet mask is 255.255.255.0. 0930155455|app1 |3|00| Gateway address is 192.168.222.1. 0930155455|app1 |3|00| DNS server is 192.168.222.4. 0930155455|app1 |3|00| DNS domain is local. 0930155455|app1 |3|00|Bootline: eim(0,0)bootHost:flash 0930155455|e=192.168.222.202:ff00:1c20:433d5fcf h=216.135.65.62 0930155455|g=192.168.222.1 u=jonathan pw=tenn1982 0930155455|app1 |3|00|Bootline: f=0x40 tn=CircaIP 0930155700|app1 |3|00|Time has been set from pool.ntp.org(193.170.141.4). 0930155700|cfg |3|00|Image bootrom.ld has not changed. 0930155701|cfg |3|00|0004f2022609.cfg could not be downloaded, getting next file. 0930155704|cfg |3|00|Image sip.ld has not changed. 0930155734|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 0930155734|app1 |6|00|Uploading boot log, time is FRI SEP 30 15:57:34 0930155734|2005 Any ideas? I attached the config files, I got them from somewhere else. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] (no subject)
Jonathan k. Creasy wrote: 0930155701|cfg |3|00|0004f2022609.cfg could not be downloaded, getting next file. Any ideas? I attached the config files, I got them from somewhere else. The phone isn't finding the config file as the above log entry shows. The config file consists of the mac address of the phone with a .cfg appended. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] (no subject)
When I receive voicemail notify via e-mail I would like receive not the phone-number, but the sender name. Where can I configure this and how? Is it possible to have some example? Thank Luca ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] (no subject)
Hi. I am using the Flash operator panel 0.24 and it works, but I don't see the voicemail icon when I have incoming voicemail. In the op_buttons.cfg I have the following setup: [SIP/100] Position=2 Label="Office tel. 1" Extension=100 Icon=1 Mailbox=100 I've tried to google on the subject, but have not found any answers. I've tried [EMAIL PROTECTED] also. (full = the context for the 100 extension)Is full the voicemail context or the extensions.conf context? If youare using a standard asterisk setup, the mailbox should probably be[EMAIL PROTECTED] (being default the voicemail context for that extension,as specified in voicemail.conf)Regards, -- Nicolás GudiñoBuenos Aires - Argentina Thank's . I put [EMAIL PROTECTED] in and it worked. Fredrik ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] (no subject)
hi all, i have a box with a te110p and a pbx siemens... connect both with a e1. with a xten soft i can call extensions numbers in my office example 100 102 etc. but when i truy to go outside with the 9 before the call rings in the first extensions (100). this is a asterisk problem? or a pbx problem? -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] (no subject)
It could potentially be both. I would look at your extensions.conf first
though. What does the extension entry for that context look like.
For instance I have an entry in my extensions.conf for dialing outside
lines (outside being from asterisk to my PBX and then onto the outside
world from there). The entry looks like this:
[to-analog]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN})
exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
To dial a PBX extension the entry would look almost the same:
[to-pbx-extension]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1})
exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
Hope this helps,
-Matt
On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
hi all, i have a box with a te110p and a pbx siemens... connect both
with a e1.
with a xten soft i can call extensions numbers in my office example 100
102 etc. but when i truy to go outside with the 9 before the call rings
in the first extensions (100). this is a asterisk problem? or a pbx
problem?
___
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