RE: [Asterisk-Users] PSTN-GATEWAY
Well the simplest is to make the connection insecure with a static ip. sip.conf [cisco2600] host=xxx.xxx.xxx.xxx defaultip=xxx.xxx.xxx.xxx insecure=yes type=friend disallow=all allow= (your codecs) extensions.conf [default] ;dial out cisco exten = _1X.,1,Dial([EMAIL PROTECTED]) As far as your cisco config for voice, depends on hardware and how you want to setup your dialpeers. ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office 813.864.3161x107 Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Reli Loin Sent: Wednesday, September 28, 2005 9:35 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PSTN-GATEWAY Hello, I have just installed asterisk and I would like to connect it to the PSTN. I have a gateway Cisco 2600, how must I declare it in the file of configuration (extensions.conf, sip.conf). thanks for your helping ___ --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 This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --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] PSTN-GATEWAY
Well the simplest is to make the connection insecure with a static ip. sip.conf [cisco2600] host=xxx.xxx.xxx.xxx defaultip=xxx.xxx.xxx.xxx insecure=yes type=friend disallow=all allow= (your codecs) extensions.conf [default] ;dial out cisco exten = _1X.,1,Dial(SIP/[EMAIL PROTECTED]) Sorry forgot the SIP/ part. :) As far as your cisco config for voice, depends on hardware and how you want to setup your dialpeers. ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office 813.864.3161x107 Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Reli Loin Sent: Wednesday, September 28, 2005 9:35 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PSTN-GATEWAY Hello, I have just installed asterisk and I would like to connect it to the PSTN. I have a gateway Cisco 2600, how must I declare it in the file of configuration (extensions.conf, sip.conf). thanks for your helping ___ --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 This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --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] PSTN Gateway X101P
Marty Mastera [EMAIL PROTECTED] wrote: When I call the pstn number, the zaptel picks up the line on the first ring and then forwards it to the sip phone and rings it. Is there anyway to prevent the zaptel from picking up the line until the sip phone actully answers the call. This way I could answer the phone either locally on a regular analog handset or through the sip phone. The way it is now, it only rings my phones in the house 1 time. Hey Jason, glad things are working...I think I understand your problem and the short answer is no - there isn't a way to ring the x-lite without asterisk answering the call first (if I'm wrong about this, someone please correct me!). If you call Answer before Dial then Asterisk will answer the line before calling the device/softphone. If you don't call Answer then the line will not be picked up until the user of the device (or softphone) answers the call. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN gateway implementation?
-I have a TE405P board and only one T1 worth of phone lines (24) connected to it using an Adtran TA750 channel bank. Any particular reason against using PRI from your telco? Is Asterisk capable of handling multiple incoming VoIP calls arriving from the same source (IP) or do I need to get something else to take the incoming traffic and pass it on to Asterisk? (I've read about using SER as a SIP proxy, but it's not clear to me wheather I need it or not). Can I use the OpenH.323 module to take care of the incoming VoIP traffic? Asterisk can handle multiple calls from the same IP without any worry. Your main worry is the lack of real billing since you're terminating to analog PSTN instead of using PRI -- you have no way of actually knowing if the call was answered or not, so he'll be billed on every call. I doubt you want to try and work with callprogress=yes. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN gateway implementation?
This is an upgrade from a previous system. The old one didn't handle PRI, so they had analog phone lines as trunks. Management won't invest the money right now to get a PRI circuit. Any suggestions? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Monday, July 19, 2004 4:59 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] PSTN gateway implementation? -I have a TE405P board and only one T1 worth of phone lines (24) connected to it using an Adtran TA750 channel bank. Any particular reason against using PRI from your telco? Is Asterisk capable of handling multiple incoming VoIP calls arriving from the same source (IP) or do I need to get something else to take the incoming traffic and pass it on to Asterisk? (I've read about using SER as a SIP proxy, but it's not clear to me wheather I need it or not). Can I use the OpenH.323 module to take care of the incoming VoIP traffic? Asterisk can handle multiple calls from the same IP without any worry. Your main worry is the lack of real billing since you're terminating to analog PSTN instead of using PRI -- you have no way of actually knowing if the call was answered or not, so he'll be billed on every call. I doubt you want to try and work with callprogress=yes. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Gateway X101P
try puttin this in extensions.conf
[outgoing]
exten = _0.,1,Dial,Zap/1/${EXTEN:1}
exten = _0.,2,Hangup
and into your siphones extensions definition
[sip]
include = outgoing
Adrià Vidal
[EMAIL PROTECTED] | http://adria.homeip.net | MSN
[EMAIL PROTECTED]
iChat [EMAIL PROTECTED] | FWD [EMAIL PROTECTED] | IAXTEL 1700 337 68
48
On Jul 18, 2004, at 5:12 PM, Jason Armentrout wrote:
1 channels configured.
It appears that I have the driver loaded correctly.
I edited the sample extensions.conf and changed the varible trunk to
zap/1
Attached is my extensions.conf
When I dial 94341321 or 4341321 I just get a 404 error in Xlite.
What am I doing wrong? Any help would be appreciated.
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Re: [Asterisk-Users] PSTN Gateway X101P
I added
exten = _0.,1,Dial,Zap/1/${EXTEN:1}
exten = _0.,2,Hangup
to the extensions.conf
but I am not sure I follow you on the second part, do you want me to add
include = outgoing
to my sip.conf file?? I did both of these changes, and I still have the same
problem.
Quoting Adria Vidal [EMAIL PROTECTED]:
try puttin this in extensions.conf
[outgoing]
exten = _0.,1,Dial,Zap/1/${EXTEN:1}
exten = _0.,2,Hangup
and into your siphones extensions definition
[sip]
include = outgoing
Adrià Vidal
[EMAIL PROTECTED] | http://adria.homeip.net | MSN
[EMAIL PROTECTED]
iChat [EMAIL PROTECTED] | FWD [EMAIL PROTECTED] | IAXTEL 1700 337 68
48
On Jul 18, 2004, at 5:12 PM, Jason Armentrout wrote:
1 channels configured.
It appears that I have the driver loaded correctly.
I edited the sample extensions.conf and changed the varible trunk to
zap/1
Attached is my extensions.conf
When I dial 94341321 or 4341321 I just get a 404 error in Xlite.
What am I doing wrong? Any help would be appreciated.
___
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Re: [Asterisk-Users] PSTN Gateway X101P
On Jul 18, 2004, at 5:56 PM, Jason Armentrout wrote: to the extensions.conf but I am not sure I follow you on the second part, do you want me to add include = outgoing to my sip.conf file?? I did both of these changes, and I still have the same problem. must add include = outgoing into your extensions.conf file where the sip extensions are defined example [sip] ; include = fwd include = iaxtel include = stanaphone include = SIPphone include = fromiaxfwd include = from-iaxtel include = stana-incoming include = parkedcalls include = outgoing exten = 100,1,Dial(SIP/100,20,tr) exten = 100,2,Voicemail,100 exten = 100,3,Hangup Adrià Vidal [EMAIL PROTECTED] | http://adria.homeip.net | MSN [EMAIL PROTECTED] iChat [EMAIL PROTECTED] | FWD [EMAIL PROTECTED] | IAXTEL 1700 337 68 48 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN Gateway X101P
What I am NOT able to do is dial a seven digit local or 10 digit long distance number and make a phone call to the pstn using the x100p card. snip Attached is my extensions.conf When I dial 94341321 or 4341321 I just get a 404 error in Xlite. What am I doing wrong? Any help would be appreciated. Hey Jason In your extensions.conf, the [default] context only has the [demo] context included which provides no outbound dialing. Try adding an 'include =' line to your default context to allow for this. For example in extensions.conf, there is a context called [local] to allow for outbound dialing, so add 'include = local' under your [default] context... The other side of this is in sip.conf, where you tell the phone (or x-lite or whatever) which context to start in (from extensions.conf). Since you can already dial 1000 and get the demo, I assume that your sip.conf is configured to start in the [default] context in extensions.conf With that being the case, after adding the include = local to your [default] context, you should be able to dial your 7 digit number (you must dial 9 first). Marty ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN Gateway X101P
Thanks for the tip, that made things work, it is really difficult for me to understand the different config files and especially the extensions.conf, it is very confusing. I am trying to learn though. Now that I have got outgoing calls to work from the sip phone. How can I route incoming calls on the pstn line (x100p) to the sip phone? Thanks! Quoting Marty Mastera [EMAIL PROTECTED]: What I am NOT able to do is dial a seven digit local or 10 digit long distance number and make a phone call to the pstn using the x100p card. snip Attached is my extensions.conf When I dial 94341321 or 4341321 I just get a 404 error in Xlite. What am I doing wrong? Any help would be appreciated. Hey Jason In your extensions.conf, the [default] context only has the [demo] context included which provides no outbound dialing. Try adding an 'include =' line to your default context to allow for this. For example in extensions.conf, there is a context called [local] to allow for outbound dialing, so add 'include = local' under your [default] context... The other side of this is in sip.conf, where you tell the phone (or x-lite or whatever) which context to start in (from extensions.conf). Since you can already dial 1000 and get the demo, I assume that your sip.conf is configured to start in the [default] context in extensions.conf With that being the case, after adding the include = local to your [default] context, you should be able to dial your 7 digit number (you must dial 9 first). Marty ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN Gateway X101P
Thanks for the tip, that made things work, it is really difficult for me to understand the different config files and especially the extensions.conf, it is very confusing. I am trying to learn though. Now that I have got outgoing calls to work from the sip phone. How can I route incoming calls on the pstn line (x100p) to the sip phone? Thanks! First, I would dial the telephone number of the line plugged into the X101P and make sure that the demo answers to verify that things are working correctly...assuming that works, you just need to modify your extensions.conf a little bit... Your [default] context includes [demo] which has an answer line in it, followed by the rest of the items necessary to playback the demo. So if you want an incoming call to ring directly to your x-lite, I would remove the include for [demo] from your [default] context (but leave the include for [local] so that you can make outbound calls!...then inside your [default] context (just below the include for [local] for example) add lines that will answer the phone and ring your x-lite: (note that below, the SIP/1000 is just an example...the '1000' should be whatever name you gave your x-lite in sip.conf) exten = s,1,Wait exten = s,2,Answer exten = s,3,Dial(SIP/1000,20,r) Save the changes and reload asterisk, try calling the line connected to the X101P and if your x-lite has registered with asterisk correctly, it should ring there...look on the wiki (www.voip-info.org) for the specific syntax of the Dial command and it's options, also the above is a very basic config, with no timeouts specified, etc...it should work, but should/could be made more robust after you get it working initially. Marty ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN Gateway X101P
Thanks Marty, That works now, the caller id on Xlite only shows the name for some reason, not the number, but anyway it now rings in. When I call the pstn number, the zaptel picks up the line on the first ring and then forwards it to the sip phone and rings it. Is there anyway to prevent the zaptel from picking up the line until the sip phone actully answers the call. This way I could answer the phone either locally on a regular analog handset or through the sip phone. The way it is now, it only rings my phones in the house 1 time. Jason Quoting Marty Mastera [EMAIL PROTECTED]: Thanks for the tip, that made things work, it is really difficult for me to understand the different config files and especially the extensions.conf, it is very confusing. I am trying to learn though. Now that I have got outgoing calls to work from the sip phone. How can I route incoming calls on the pstn line (x100p) to the sip phone? Thanks! First, I would dial the telephone number of the line plugged into the X101P and make sure that the demo answers to verify that things are working correctly...assuming that works, you just need to modify your extensions.conf a little bit... Your [default] context includes [demo] which has an answer line in it, followed by the rest of the items necessary to playback the demo. So if you want an incoming call to ring directly to your x-lite, I would remove the include for [demo] from your [default] context (but leave the include for [local] so that you can make outbound calls!...then inside your [default] context (just below the include for [local] for example) add lines that will answer the phone and ring your x-lite: (note that below, the SIP/1000 is just an example...the '1000' should be whatever name you gave your x-lite in sip.conf) exten = s,1,Wait exten = s,2,Answer exten = s,3,Dial(SIP/1000,20,r) Save the changes and reload asterisk, try calling the line connected to the X101P and if your x-lite has registered with asterisk correctly, it should ring there...look on the wiki (www.voip-info.org) for the specific syntax of the Dial command and it's options, also the above is a very basic config, with no timeouts specified, etc...it should work, but should/could be made more robust after you get it working initially. Marty ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN Gateway X101P
When I call the pstn number, the zaptel picks up the line on the first ring and then forwards it to the sip phone and rings it. Is there anyway to prevent the zaptel from picking up the line until the sip phone actully answers the call. This way I could answer the phone either locally on a regular analog handset or through the sip phone. The way it is now, it only rings my phones in the house 1 time. Jason Hey Jason, glad things are working...I think I understand your problem and the short answer is no - there isn't a way to ring the x-lite without asterisk answering the call first (if I'm wrong about this, someone please correct me!). It sounds like your analog telephone isn't connected into the asterisk box, but instead just plugged into a standard wall outlet somewhere, connected directly to the pstn. If this is the case, you will be limited b/c asterisk must answer the call before it can do any other processing such as ring another phone, etc...you might be able to configure asterisk to answer after 5 rings or something, giving you a chance to answer the analog phone first, but most people would probably do the following: The way around this is to connect your analog phone into asterisk and have asterisk ring the analog phone and the x-lite simultaneously, giving you the choice of how to answer it. There are a couple of ways to do this, such as a Digium TDM400B pci card with 1 FXS module installed in it (to which you would connect the phone), or a SIP (or H.323, or IAX) to FXS adapter such as the cisco ata 286 or the sipura 2000, etc.. (various models are described on the wiki)... There are plenty of advantages to this such as music on hold, the ability to transfer calls between x-lite and the analog phone, and plenty more as described on the wiki.. Marty ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN gateway
- Original Message - From: Deepakumar JV [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 8:37 AM Subject: [Asterisk-Users] PSTN gateway Hello Has anyone come across a small residential PSTN gateway? Its not worth running a * just as a PSTN gateway as it requries a seperate system / power / etc... I am looking for a device that could connect to * and a pstn line so that i could register that device to * and make pstn calls via that device. I'm confused. Do you want to get rid of *, or not? It sounds like you're just looking for an IP phone to pstn gateway service. See: vonage, voicepulse, etc... - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN gateway
Sorry for confusing.. let me explain ideally i want to have two * running, one at my place and the other at a remote location. Now the problem in running * at a remote location is the effort / cost involved in setting up / maintaining the * box. Hence i was looking for a device that could register with * (as a client so that i could dial a number and reach it as a normal extension) and also have a PSTN connectivity at the remote location. The reason i need PSTN connectivity at remote location is to make outbound calls from * via the device so called PSTN gateway. If i am still not clear, then in simple terms, i am looking for a hardware device with one FXO port and SIP support. Any help or suggestion please Thanks in advance Deepak - Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 01:58 PM Subject: Re: [Asterisk-Users] PSTN gateway - Original Message - From: Deepakumar JV [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 8:37 AM Subject: [Asterisk-Users] PSTN gateway Hello Has anyone come across a small residential PSTN gateway? Its not worth running a * just as a PSTN gateway as it requries a seperate system / power / etc... I am looking for a device that could connect to * and a pstn line so that i could register that device to * and make pstn calls via that device. I'm confused. Do you want to get rid of *, or not? It sounds like you're just looking for an IP phone to pstn gateway service. See: vonage, voicepulse, etc... - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN gateway
Hi! Has anyone come across a small residential PSTN gateway? It's all there: http://www.voip-info.org/wiki-VOIP+Phones Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
