Re: [asterisk-users] How Secure Is Asterisk
On Mon, 2008-10-20 at 14:01 -0500, Steve Anness wrote: I am sure this has been discussed prior, however, I am sitting here and being asked this very question by my superiors. Ahh stuperiors, don't you love the questions they ask? Almost as good as the questions some recruiters (by this I mean the people who normally recruit accountants or secretaries and think they can effectivly recruit IT staff) ask. They are loving what I have done with our two Asterisk servers here; however, they keep asking me if it is secure or not. Of course, as with anything, I suspect that on a secure network they can be reasonably safe. Are you after security of the host? the client? the application? or of the data being transmitted? Depending on how you are making * available and what you are after the network may play a role in making things secure. However, realistically if I am using the asterisk server to make internal calls and discussion very private matters, how possible is it for someone to listen to calls? How good is the encryption if any over an IAX trunk? There is no encryption on SIP or IAX. If you are only making internal calls (i.e. there is no external exposure of *) then you could put the phones and the server on their own physical [or virtual] LAN and restrict access on this [V]LAN to known mac addresses (so just known IP phones), this would help with the security of conversations ... it's also worth noting that most decent modern switches will make it very difficult to eavesdrop on a network connection that is not destined for the listening host. As has been mentioned if you were able to run some kind of VPN connection to the phones this would also be another step towards security. Some of this will also come down to your dialplan and what you let clients getaway with. If the server is facing a public network you might want to stick a firewall in front of it. The security of any application or solution is something that is dependant on many separate, sometimes overlapping issues and is something that is always changing. In this case I would be looking at your network design and the configuration of * in total, but especially the dialplan. -- Nikolai Lusan [EMAIL PROTECTED] ___ -- 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] sip and nat
hi there, I 'm a newbie in VOIP technologies ; i 'm implementing asterisk and i 'm wonder what is the best way to resolving the Asterisk/NAT problem : some clients are behind a NAT. anyone could help me? thanks johanna _ Appelez vos amis de PC à PC -- C'EST GRATUIT http://get.live.com/messenger/overview ___ -- 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] sip and nat
hi there, I 'm a newbie in VOIP technologies ; i 'm implementing asterisk and i 'm wonder what is the best way to resolving the Asterisk/NAT problem : some clients are behind a NAT. anyone could help me? thanks johanna _ Appelez vos amis de PC à PC -- C'EST GRATUIT http://get.live.com/messenger/overview ___ -- 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] sip and nat
John, Client Behind a NAT should not be problem. What are your issues? If you post your scenario and more details about your problem only then some can help you better. Jai Buy SIP DID at www.didforsale.com On Wed, Oct 22, 2008 at 12:24 AM, Johanna NIRINA [EMAIL PROTECTED]wrote: hi there, I 'm a newbie in VOIP technologies ; i 'm implementing asterisk and i 'm wonder what is the best way to resolving the Asterisk/NAT problem : some clients are behind a NAT. anyone could help me? thanks johanna _ Appelez vos amis de PC à PC -- C'EST GRATUIT http://get.live.com/messenger/overview ___ -- 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] sip and nat
I'm using asterisk 1.4 . There is some sip clients is behind a NAT : the asterisk server can't send request to these client. I'm looking for a solution to solve that in the server (asterisk) side. (sorry for my english). thanks, johanna _ Découvrez Windows Live Spaces et créez votre site Web perso en quelques clics ! http://spaces.live.com/signup.aspx ___ -- 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] How Secure Is Asterisk
On 20 Oct 2008, at 20:01, Steve Anness wrote: I am sure this has been discussed prior, however, I am sitting here and being asked this very question by my superiors. They are loving what I have done with our two Asterisk servers here; however, they keep asking me if it is secure or not. Of course, as with anything, I suspect that on a secure network they can be reasonably safe. However, realistically if I am using the asterisk server to make internal calls and discussion very private matters, how possible is it for someone to listen to calls? How good is the encryption if any over an IAX trunk? The IAX encryption (encryption=yes in iax.conf) is actually pretty good from what I can see. 3 things though: 1) you can't tell if it has happened - if the far end changes config to encryption=no nothing breaks, your calls just go through un-encrypted - I'd like a must_encrypt setting. 2) The keys are as strong as your iax passwords and the quality of / dev/random on your box. 3) The dialed number, caller id etc all go in the clear, the call setup is unencrypted. Only the body of the call is covered by the encryption. Also there are _no_ endpoints that implement it (except asterisk and our phonefromhere.com softphone) so the last yards to your user will not be protected. Tim. ___ -- 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] How Secure Is Asterisk
On 22 Oct 2008, at 07:23, Nikolai Lusan wrote: On Mon, 2008-10-20 at 14:01 -0500, Steve Anness wrote: However, realistically if I am using the asterisk server to make internal calls and discussion very private matters, how possible is it for someone to listen to calls? How good is the encryption if any over an IAX trunk? There is no encryption on SIP or IAX. If you are only making internal calls (i.e. there is no external exposure of *) then you could put the There is in IAX - set encryption=yes in iax.conf at both ends of a link and all md5 auth'd IAX calls between them will be AES encrypted. Very cool, very easy. Tim ___ -- 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] WebCall application
On 22 Oct 2008, at 10:44, voip crazy wrote: Hello list, Does anybody know any free WebCall solution to let our customer call us directly via our web site? Any clue will be welcomed. Yep, take a look at our offering on www.phonefromhere.com Tim. ___ -- 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] WebCall application
Hello list, Does anybody know any free WebCall solution to let our customer call us directly via our web site? Any clue will be welcomed. Thanks. VoipCrazy ___ -- 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] 2 asterisk boxes
I am trying to setup a second asterisk box to play with console/dsp over sip. My sip.conf on the second box is: [secondbox] type=friend username=secondbox secret=secret disallow=all allow=ulaw allow=alaw allow=gsm host=SERVERIP context=consoledsp The second box is not connecting to my asterisk server. When I startup asterisk and I enter sip set debug I never see anything being displayed... sip show peers on the second box shows: sip show peers Name/username HostDyn Nat ACL Port Status secondbox SERVERIP 5060 Unmonitored 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline] However I never see connection attempts, I dont see anything being logged that its failing to connect. Sip show peers on the server has: secondbox (Unspecified)D 0Unmonitored running sip set debug on the server I never see a connection attempt from the secondbox to look at any error messages why its not connecting. I have done a service iptables stop on the second box. The server is OK as it has phones on it. How do I tell why the secondbox is not connecting to the server??? Thanks, Jerry ___ -- 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] Asterisk Console color
-Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] På vegne av Dwayne Hubbard Sendt: 21. oktober 2008 19:04 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Asterisk Console color - Armand Fumal [EMAIL PROTECTED] wrote: Since I'm using ubuntu 8.04.1 and asterisk 1.4.22 I cannot have color in console. Do I miss a package or compilation option ? Edit your terminal foreground and background color scheme to white on black -Dwayne. And also change Built-In-Schemes to XTerm. That worked for me... -- Ivar ___ -- 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] WebCall application
Tim Panton wrote: Does anybody know any free WebCall solution to let our customer call us directly via our web site? Any clue will be welcomed. Yep, take a look at our offering on www.phonefromhere.com A per-minute charge does not constitute a free solution. Please read requests before spouting off about your own products. ___ -- 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] WebCall application
On 22 Oct 2008, at 14:28, Rob Hillis wrote: Tim Panton wrote: Does anybody know any free WebCall solution to let our customer call us directly via our web site? Any clue will be welcomed. Yep, take a look at our offering on www.phonefromhere.com A per-minute charge does not constitute a free solution. Please read requests before spouting off about your own products. oops, sorry, take a look at this instead http://code.google.com/p/njiax/ Of course someone still pays for the bandwidth but. Tim. ___ -- 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] 2 asterisk boxes
Jerry Geis wrote: I am trying to setup a second asterisk box to play with console/dsp over sip. My sip.conf on the second box is: [secondbox] type=friend username=secondbox secret=secret disallow=all allow=ulaw allow=alaw allow=gsm host=SERVERIP context=consoledsp The second box is not connecting to my asterisk server. When I startup asterisk and I enter sip set debug I never see anything being displayed... sip show peers on the second box shows: sip show peers Name/username HostDyn Nat ACL Port Status secondbox SERVERIP 5060 Unmonitored 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline] However I never see connection attempts, I dont see anything being logged that its failing to connect. Sip show peers on the server has: secondbox (Unspecified)D 0Unmonitored running sip set debug on the server I never see a connection attempt from the secondbox to look at any error messages why its not connecting. I have done a service iptables stop on the second box. The server is OK as it has phones on it. How do I tell why the secondbox is not connecting to the server??? Thanks, Jerry ___ -- 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 Take a look at http://www.voip-info.org/wiki-Asterisk+config+sip.conf specifically the section labeled Asterisk as a sip client -- Robin D. Rodriguez Systems Engineer Ifbyphone, Inc. Phone: (866) 250-1663 Fax: (847) 676-6553 [EMAIL PROTECTED] http://www.ifbyphone.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] sip and nat
Johanna NIRINA wrote: I'm using asterisk 1.4 . There is some sip clients is behind a NAT : the asterisk server can't send request to these client. I'm looking for a solution to solve that in the server (asterisk) side. (sorry for my english). thanks, johanna _ Découvrez Windows Live Spaces et créez votre site Web perso en quelques clics ! http://spaces.live.com/signup.aspx ___ -- 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 Common solutions include stun or a combination of qualify=yes and/or nat=yes entries in sip.conf http://www.voip-info.org/wiki/view/Asterisk+sip+qualify -- Robin D. Rodriguez Systems Engineer Ifbyphone, Inc. Phone: (866) 250-1663 Fax: (847) 676-6553 [EMAIL PROTECTED] http://www.ifbyphone.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] Asterisk Console color
On Tue, Oct 21, 2008 at 06:43:31PM +0200, Armand Fumal wrote: Hi, Since I'm using ubuntu 8.04.1 and asterisk 1.4.22 I cannot have color in console. Do I miss a package or compilation option ? http://bugs.digium.com/9048 The fix for this issue is to tell Asterisk to pretend it has a valid terminal if it is daemonized. It doesn't matter for the daemon either way, and it does matter for remote terminals. -- Tzafrir Cohen icq#16849755 jabber:[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] Parking Issue
HI all, I have a question, is call parking broken: When you park a call it says it will time out to a certain extension in a certain context, it never does it just calls the parker back. How do you get it to timeout to certain extension? -- Executing [EMAIL PROTECTED]:2] Park(SIP/testing-b7701418, ) in new stack == Parked SIP/testing-b7701418 on [EMAIL PROTECTED] Will timeout back to extension [craigp] s, 1 in 10 seconds -- SIP/testing-b7701418 Playing 'digits/7' (language 'en') -- SIP/testing-b7701418 Playing 'digits/0' (language 'en') -- SIP/testing-b7701418 Playing 'digits/1' (language 'en') -- Added extension '71' priority 1 to parkedcalls -- Started music on hold, class 'default', on channel 'SIP/testing-b7701418' == Spawn extension (craigp, s, 1) exited KEEPALIVE on 'SIP/testing-b7701418' Thanks, Craig ___ -- 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] hex b1 in CallerID sent by Asterisk On PRI
On Tue, 2008-10-21 at 13:56 -0500, Bob Pierce wrote: Does anyone know what the significance is of the b1 being sent here? Or, is there a way to make Asterisk not send the b1 character as a test? As an update to this, I noticed the following lines in libpri.h near line 236: /* Network Specific Facilities (ATT) */ #define PRI_NSF_NONE -1 #define PRI_NSF_SID_PREFERRED 0xB1 #define PRI_NSF_ANI_PREFERRED 0xB2 So, I've tried specifying nsf=none in zapata.conf, but we still see the b1 preceding the caller name in the pri trace. However, I only did a 'reload chan_zap.so' since this is a production system. Should that have changed the nsf settings for this span? My current zapata.conf is pasted below if that helps... Bob [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A104 port 1 [slot:4 bus:13 span:1] wanpipe1 switchtype=national context=inbound group=1 signalling=pri_cpe channel =1-23 ;Sangoma A104 port 2 [slot:4 bus:13 span:2] wanpipe2 context=stations group=0 signalling=fxo_ks channel = 25-48 ;Sangoma A104 port 3 [slot:4 bus:13 span:3] wanpipe3 switchtype=national nsf=none context=metaswitch group=2 signalling=pri_cpe channel =49-71 ___ -- 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] fax / t38 gateway
I'm trying to figure out how to handle our fax line when we switch to our asterisk for voice. After a lot of reading and poking about I have concluded, as have many others it would seem, that the best thing to do is either to have a separate pstn fax line or use some sort of internet faxing service rather than try and make faxing work in a way it's not meant to over voip lines. The question I can't seem to find a good answer to is if there is a service/software that would allow a DID to be transferred to them and then they perform the t.38 gateway/conversion functions to which I can connect with asterisk as a t.38 endpoint and originator, or if there is a way that I could host that on my own server? So essentially I am a bit confused that asterisk supports t.38 as an endpoint or originator, but there doesn't seem to be a way to convert to/from analog for interoperating with normal fax machines. I'm sure something exists or the code wouldn't have been written into asterisk... Can someone point me in the right direction? Brendan Martens ___ -- 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] asterisk video
hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start troubleshooting this? TIA Regards nhadie ___ -- 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] Sonicwall potentially causing long ping times to SIP phones
Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on sip show peer shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify? Thanks. -- James ___ -- 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] How to add contexts in asterisk realtime?
hi for any context ,you must to open /etc/asterisk/extensions.conf and insert this line : exten =Realtime/[EMAIL PROTECTED] and (reload) or (restart now) your asterisk You don't have to restart asterisk, just a 'dialplan reload' will suffice. So really there is no impact to a running system. JR -- - JR Richardson Engineering for the Masses ___ -- 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] fax / t38 gateway
What version of *? Are you going all VOIP for your voice or are you using a T1/E1? *? 1.4 has t38 pass-through and 1.6 has pass-through and termination, but 1.6 was just release and I would not suggest using it in a production environment unless you can tolerate problem or even outages. If you are planning on using a T1/E1 then send incoming calls to iaxmodem/hylafax or to an ATA/FXS card. Either works very well. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brendan Martens Sent: Wednesday, October 22, 2008 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] fax / t38 gateway I'm trying to figure out how to handle our fax line when we switch to our asterisk for voice. After a lot of reading and poking about I have concluded, as have many others it would seem, that the best thing to do is either to have a separate pstn fax line or use some sort of internet faxing service rather than try and make faxing work in a way it's not meant to over voip lines. The question I can't seem to find a good answer to is if there is a service/software that would allow a DID to be transferred to them and then they perform the t.38 gateway/conversion functions to which I can connect with asterisk as a t.38 endpoint and originator, or if there is a way that I could host that on my own server? So essentially I am a bit confused that asterisk supports t.38 as an endpoint or originator, but there doesn't seem to be a way to convert to/from analog for interoperating with normal fax machines. I'm sure something exists or the code wouldn't have been written into asterisk... Can someone point me in the right direction? Brendan Martens ___ -- 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] Sonicwall potentially causing long ping times toSIP phones
Sonicwalls from the TZ line and before line do seem to have a number of issues with VoIP. Jeff Johnson Director of Operations NeturallySpeaking, LLC sip://[EMAIL PROTECTED] http://www.neturallyspeaking.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Lamanna Sent: Wednesday, October 22, 2008 2:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sonicwall potentially causing long ping times toSIP phones Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on sip show peer shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify? Thanks. -- James ___ -- 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 No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.173 / Virus Database: 270.8.2/1737 - Release Date: 10/21/2008 9:10 AM This email and any attached files are confidential and intended solely for the intended recipient(s). If you are not the named recipient you should not read, distribute, copy or alter this email. Any views or opinions expressed in this email are those of the author and do not represent those of the company. Warning: Although precautions have been taken to make sure no viruses are present in this email, the company cannot accept responsibility for any loss or damage that arise from the use of this email or attachments. ___ -- 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] Sonicwall potentially causing long ping times toSIP phones
I had weird issues when using a Sonicwall, gave up. Stuck in linksys running dd-wrt firmware running on a separate VLAN... no issues since -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Lamanna Sent: Wednesday, October 22, 2008 12:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sonicwall potentially causing long ping times toSIP phones Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on sip show peer shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify? Thanks. -- James ___ -- 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] changing from default codec
hi, using sip, my default codec is set to gsm in sip.conf I occasionally want to send out a call using ulaw while other channels are using gsm, how can I do this using call files ? I couldn't find any codec parameter in the call file definition. tia. ___ -- 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] : Parking Issue
HI all, I have a question, is call parking broken: When you park a call it says it will time out to a certain extension in a certain context, it never does it just calls the parker back. How do you get it to timeout to certain extension? -- Executing [EMAIL PROTECTED]:2] Park(SIP/testing-b7701418, ) in new stack == Parked SIP/testing-b7701418 on [EMAIL PROTECTED] Will timeout back to extension [craigp] s, 1 in 10 seconds -- SIP/testing-b7701418 Playing 'digits/7' (language 'en') -- SIP/testing-b7701418 Playing 'digits/0' (language 'en') -- SIP/testing-b7701418 Playing 'digits/1' (language 'en') -- Added extension '71' priority 1 to parkedcalls -- Started music on hold, class 'default', on channel 'SIP/testing-b7701418' == Spawn extension (craigp, s, 1) exited KEEPALIVE on 'SIP/testing-b7701418' Thanks, Craig ___ -- 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] asterisk video
On Wednesday 22 October 2008 12:27:17 Nhadie wrote: hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start troubleshooting this? Uncomment videosupport=yes in sip.conf. -- Tilghman ___ -- 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] How to add contexts in asterisk realtime?
hi for any context ,you must to open /etc/asterisk/extensions.conf and insert this line : exten =Realtime/[EMAIL PROTECTED] and (reload) or (restart now) your asterisk You don't have to restart asterisk, just a 'dialplan reload' will suffice. So really there is no impact to a running system. You've obviously never tried doing that on a system with 50,000+ extensions and having to reload every time a new customer signs up via an online web interface... ___ -- 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] Sonicwall potentially causing long ping timestoSIP phones
Sonicwalls TZ170 and older have issues with SIP Jeff Johnson Director of Operations NeturallySpeaking, LLC sip://[EMAIL PROTECTED] http://www.neturallyspeaking.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Van Ham Sent: Wednesday, October 22, 2008 3:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Sonicwall potentially causing long ping timestoSIP phones I had weird issues when using a Sonicwall, gave up. Stuck in linksys running dd-wrt firmware running on a separate VLAN... no issues since -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Lamanna Sent: Wednesday, October 22, 2008 12:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sonicwall potentially causing long ping times toSIP phones Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on sip show peer shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify? Thanks. -- James ___ -- 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 No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.173 / Virus Database: 270.8.2/1737 - Release Date: 10/21/2008 9:10 AM This email and any attached files are confidential and intended solely for the intended recipient(s). If you are not the named recipient you should not read, distribute, copy or alter this email. Any views or opinions expressed in this email are those of the author and do not represent those of the company. Warning: Although precautions have been taken to make sure no viruses are present in this email, the company cannot accept responsibility for any loss or damage that arise from the use of this email or attachments. ___ -- 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] hex b1 in CallerID sent by Asterisk On PRI
On Wed, 2008-10-22 at 12:11 -0500, Bob Pierce wrote:
On Tue, 2008-10-21 at 13:56 -0500, Bob Pierce wrote:
Does anyone know what the significance is of the b1 being sent here?
Or, is there a way to make Asterisk not send the b1 character as a
test?
As a further update to this, I've noticed the following in q931.c at
about line 1236:
static FUNC_SEND(transmit_display)
{
int i;
if ((pri-switchtype == PRI_SWITCH_QSIG) ||
((pri-switchtype == PRI_SWITCH_EUROISDN_E1) (pri-localtype ==
PRI_CPE)) ||
!call-callername[0])
return 0;
i = 0;
if(pri-switchtype != PRI_SWITCH_EUROISDN_E1) {
ie-data[0] = 0xb1;
++i;
}
memcpy(ie-data + i, call-callername, strlen(call-callername));
return 2 + i + strlen(call-callername);
}
So, I think this is where the b1 is being added.
My question then is, what is the significance of this character?
What's the best way to try sending caller name without this character?
Should I just try changing my switchtype to euroisdn at both sides of
the link?
Bob
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Re: [asterisk-users] : Parking Issue
On 22 Oct 2008, at 20:29, Craig Van Ham wrote: HI all, snip This appears to be the same message you posted earlier. ___ -- 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] adding a second extension
I also tried downgrading to version 1.4-current but that didn't help. Any other ideas? I'm at a loss. ___ -- 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] adding a second extension
What kind of phone are you trying to connect to 101??? and from where? On Wed, Oct 22, 2008 at 7:07 PM, Stephen Reese [EMAIL PROTECTED] wrote: I also tried downgrading to version 1.4-current but that didn't help. Any other ideas? I'm at a loss. -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227 ___ -- 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] asterisk video
hi sir, i uncommented that as mentioned on the howto. regards, nhadie Tilghman Lesher wrote: On Wednesday 22 October 2008 12:27:17 Nhadie wrote: hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start troubleshooting this? Uncomment videosupport=yes in sip.conf. ___ -- 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] adding a second extension
On Wed, Oct 22, 2008 at 8:15 PM, Juan Rodríguez [EMAIL PROTECTED] wrote: What kind of phone are you trying to connect to 101??? and from where? Both phones are Cisco, 101 is a 7960 and 102 is a 7912. 101 can contact 102 by dialing 101 but not the other way around, I just get a busy tone. ___ -- 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] FW: [wwwac] Thursday 23 October 2008 NYLUG: Paul Charles Leddy on Asterisk, the Free Software Telephone System
I hadn't seen anything on the asterisk list but just in case anyone is interest. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: Murat Aktihanoglu [mailto:[EMAIL PROTECTED] Sent: Wednesday, 22 October 2008 9:57 PM To: Dean Collins Subject: Fwd: [wwwac] Thursday 23 October 2008 NYLUG: Paul Charles Leddy on Asterisk, the Free Software Telephone System Hi Dean, thought you might be interested in this, cheers, Murat http://unype.com http://unype.com/blog 917 656 9309 -- Forwarded message -- From: Jay Sulzberger [EMAIL PROTECTED] Date: Wed, Oct 22, 2008 at 11:48 AM Subject: [wwwac] Thursday 23 October 2008 NYLUG: Paul Charles Leddy on Asterisk, the Free Software Telephone System To: [EMAIL PROTECTED] blockquote what=official NYLUG announcement edits= From: NYLUG Announcements [EMAIL PROTECTED] To: NYLUG Announcements [EMAIL PROTECTED] Date: Tue, 21 Oct 2008 16:34:20 -0400 (EDT) Subject: [nylug-announce] NYLUG Oct 23 Meeting 6:30PM, Paul Charles Leddy on a Technical Overview of Asterisk Thursday, October 23, 2008 6:30pm-8:00 PM IBM 590 Madison Ave, 12th Floor corner of 57th Street ** RSVP Closes at 4:30pm the day of the meeting (sharp!) *** Please RSVP for EVERY meeting at this time. Register at http://rsvp.nylug.org/ Check in with photo ID at the lobby for badge. Paul Charles Leddy - on - The Asterisk Free Software Telephone System Please join us Thursday October 23 for a demonstration and overview of Asterisk, a Free Software implementation of a telephone switch. What is a switch? Well, it's the essential core of a phone company. It's also what lets you dial 42 on your phone at work and be connected to the person who knows everything. Wait, that's you? Paul Charles Leddy will be guiding us through a technical overview of how Asterisk works, rather than an overview of how to use it. This talk will be of interest to anyone interested in how a PBX works, as well as those interested in building their own Asterisk PBX. We'll start with a look at the various things Asterisk does, and work our way through setting up extensions, VoIP, SIP, IAX, and call routing. More Information: * Asterisk Web Site http://www.asterisk.org/ * Alex Pilosov's Asterisk presentation in 2003 http://nylug.org/meetings/index.shtml?20030700 About Paul Charles Leddy: Paul Charles Leddy is someone who likes to make things go. He had planned on becoming an electrical engineer after Tulane, but dropped out, turned to music, rode the Internet Bubble, and then settled into a life as a Linux sysadmin. You'll find him around Portland and New York. Meeting Location: Please note that this meeting will be held at IBM, 590 Madison Ave, 12th floor, corner of 57th Street, and not at Google. This is the building with the IBM logo on the front of the building. Map: http://nylug.org/mapofibm Books!!! Our friends at Prentice-Hall kindly provide us with review copies of various new titles. One of these could be yours, all you have to do is agree to review the book within a reasonable period of time. Swag (Give Away): During/after the meeting... unusually terrific swag may be given away. Stammtisch: After the meeting ... You may wish to join up with other NYLUGgers over at TGI Fridays located at 677 Lexington Avenue and 56th Street, second floor. Northeast corner. Python Workshops: We are rounding up a group that wants to learn Python. This would be a great time to attend our workshop. The workshops meet every other Tuesday, at the NY Public Library, Hudson Park Branch. 66 Leroy St. NY NY from 6:00 PM - 8:00 PM Next meetings are October 28th followed by November 11th. See the calendar at: http://nylug.org/pythoncalendar Please see our home page at http://www.nylug.org for the HTMLized version of this announcement, our archives, and a lot of other good stuff. __ Hire expert Linux talent by posting jobs here :: http://jobs.nylug.org nylug-announce mailing list [EMAIL PROTECTED] http://nylug.org/mailman/listinfo/nylug-announce /blockquote Distributed poC TINC: Jay Sulzberger [EMAIL PROTECTED] Corresponding Secretary LXNY LXNY is New York's Free Computing Organization. http://www.lxny.org ## WWWAC Lively Chat - http://lively.com/dr?rid=5635871238148598291 ## ## The World Wide Web Artists' Consortium - http://www.wwwac.org/ ## ## To Unsubscribe, send email to: [EMAIL PROTECTED] ## ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or
Re: [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!
TDE does NOT need a VoIP card, you need to buy a DSP card, VoIP is built in. In fact if you are making pure VoIP extensions to extension calls you don't even need the DSP card/s. What type of VoIP are you trying to accomplish with Asterisk? Extensions to extensions? or Provider based. In any event I don't think it makes any difference, since adding Asterisk just for VoIP will cost the customer at least $2000.00 plus the phones plus some card in the TDE (PRI Card?) that will talk with Asterisk, while adding the DSP cards will cost way less and they will be happier since the Panasonic proprietary VoIP phones look the same as the digital proprietary phones. The only drawback with Panasonic Proprietary VoIP: you do need a VPN to every point, and if you will be using Peer to Peer (the MPR on the TDE does), each VPN endpoint must be able to communicate with every other endpoint (there is some fancy name for this but I forgot what it's called). On Tue, Oct 21, 2008 at 8:54 AM, César García [EMAIL PROTECTED] wrote: Hello Rodolfo, I see you have experience with Panasonic, and I have a new challenge of integrating Asterisk in an enterprice where they have a KX-TDE200 without the VoIP card so they can' t have voip with the pana-PBX, and that's why they want *, so do you have any advices for me :) ? I need to integrate asterisk and keep the panasonic :( because it is almos new. Thanks a lot 2008/10/17 C F [EMAIL PROTECTED] On Fri, Oct 17, 2008 at 8:02 AM, Rodolfo Alcazar Portillo [EMAIL PROTECTED] wrote: You are argueing with things like I can do it with panasonic, but it's not documented anywhere, documentation is a mess but not poor, sorry for underestimate your abilities. Sorry but I do [completely understanding Panasonic PBXs]. Not technical. Worst even: that's the propietary software culture. Many thanks for your advice, LED's issue was an imprtant feature to keep the eye on. I'll stick with asterisk, for many reasons *. Thanks again. Good luck. R * Only one (have more examples): even though I have old pana-PBXs, I bought a TDA100 (new model as provider offered). Has some SMDR bug, CERTIFIED tech tried to upgrade firmware twice, can not, ended programming it three times, costs us tens hours of service, until guarantee is lost, now works worst as initially: has noise on one line. Surely there is a fix (though not documented, as you wrote)... Both the fact tech couldn't update it and the noise indicate the tech didn't do it right. There is specific well documented procedure how to do it. The reset precess should take care of both problems. ... and bug stills strong as ever. Am Freitag, den 17.10.2008, 00:57 -0400 schrieb C F: On Thu, Oct 16, 2008 at 7:25 AM, Rodolfo Alcazar Portillo [EMAIL PROTECTED] wrote: Am Mittwoch, den 15.10.2008, 20:51 -0400 schrieb C F: Being a Panasonic dealer and having more than 50 Asterisk system in production, I can tell you that if this is your first Asterisk project, then go with Panasonic, you'll safe yourself lots of aggravation and have a happier customer. You are completely wrong! Last 4 years, I installed/programmed 6 Panasonic (KXTD1232, 3x TA308, TDA100, TEM824) in our offices. TD1232 has been discontinued for at least 5 years. Don't know about the the TA308 since the last and only one I installed was in 1998, but I have not seen them advertised in the last 5 years. Which makes me think they are discontinued as well. Until now, I don't completely understand Sorry but I do. them. Their GUI software is really bad. The functions are awfully limited. Manuals are poor. Mailing lists with helpful people there is not. GUI on the TD is really really bad. Functions are not limited, like you said: I don't completely understand them. GUI on the TDA is nice and organized. Documentation is a mess but not poor. The main reason being it's translated from Japanise, and they don't explain the theory just the steps. Yes no mailing lists. Less than a week ago (friday), bought 3FXS, 1FXO with SIP (sipura/linksys), and KNOW NOTHING ABOUT asterisk. Today, I emulated almost all features we use (account codes, DISA, own dial plans), and I can really say: ASTERISK WORKS INCREDIBLE! I even programmed an AGI script, which injects a variable to extensions.conf; on the other hand, that means I can reboot a server from my cellphone, isn't that incredible? I can do that with Panasonic as well, no it's not documented anywhere in Panasonic docs. now, I dont' know how, but 99% I'm sure I can trigger a phonecall when one server is offline. Only with asterisk. I'm almost sure a Panasonic can't emulate this features. Maybe with some expensive software. Then, I'm going to suggest 30 Voip phones, 2 8xFXO digium. I made an informal presentation yesterday, the people were amused. Thanks the people
Re: [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!
On Tue, Oct 21, 2008 at 11:55 AM, Rodolfo Alcazar Portillo [EMAIL PROTECTED] wrote: Am Dienstag, den 21.10.2008, 06:54 -0600 schrieb César García: Hello Rodolfo, I see you have experience with Panasonic, and I have a new challenge of integrating Asterisk in an enterprice where they have a KX-TDE200 without the VoIP card so they can' t have voip with the pana-PBX, and that's why they want *, so do you have any advices for me :) ? I need to integrate asterisk and keep the panasonic :( because it is almos new. Ahm, i think the best way to integrate them is one panasonic VoIP card, which supports H323V2, I think (don't have one). Don't know if TDE200 supports it. The TDE MPR supports H323 and SIP, licensing and DSP is required to make it work. I think it's better to choose one technology only to provide your service (asterisk or panasonic). Without a voip card, you are trying to connect two very different technologies, they never will be fully integrated, and the users take the worst part. The best I did is I connecting 3 PBXs on different parts of the city (tough connected with fiber opticals) with Linksys PAP2 FXS ports as trunks. :) -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft für Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gestión Pública Descentralizada y Lucha Contra La Pobreza - PADEP Av. Sánchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email: [EMAIL PROTECTED] ___ -- 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] fax / t38 gateway
I am using 1.6.0.1 and we are going to be pure voip. I know it has pass through and termination, but that is useless if I don't have a way to transform the analog t.30 to t.38 before it gets to me. That is where my confusion lays, is there some way of doing this that I am not aware of? Brendan Martens On Oct 22, 2008, at 3:02 PM, Jonn R Taylor wrote: What version of *? Are you going all VOIP for your voice or are you using a T1/E1? *? 1.4 has t38 pass-through and 1.6 has pass-through and termination, but 1.6 was just release and I would not suggest using it in a production environment unless you can tolerate problem or even outages. If you are planning on using a T1/E1 then send incoming calls to iaxmodem/hylafax or to an ATA/FXS card. Either works very well. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Brendan Martens Sent: Wednesday, October 22, 2008 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] fax / t38 gateway I'm trying to figure out how to handle our fax line when we switch to our asterisk for voice. After a lot of reading and poking about I have concluded, as have many others it would seem, that the best thing to do is either to have a separate pstn fax line or use some sort of internet faxing service rather than try and make faxing work in a way it's not meant to over voip lines. The question I can't seem to find a good answer to is if there is a service/software that would allow a DID to be transferred to them and then they perform the t.38 gateway/conversion functions to which I can connect with asterisk as a t.38 endpoint and originator, or if there is a way that I could host that on my own server? So essentially I am a bit confused that asterisk supports t.38 as an endpoint or originator, but there doesn't seem to be a way to convert to/from analog for interoperating with normal fax machines. I'm sure something exists or the code wouldn't have been written into asterisk... Can someone point me in the right direction? Brendan Martens ___ -- 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] fax / t38 gateway
If you are VoIP-only then you need a SIP provider that offers T.38. On Wed, Oct 22, 2008 at 11:17 PM, Brendan Martens [EMAIL PROTECTED] wrote: I am using 1.6.0.1 and we are going to be pure voip. I know it has pass through and termination, but that is useless if I don't have a way to transform the analog t.30 to t.38 before it gets to me. That is where my confusion lays, is there some way of doing this that I am not aware of? Brendan Martens On Oct 22, 2008, at 3:02 PM, Jonn R Taylor wrote: What version of *? Are you going all VOIP for your voice or are you using a T1/E1? *? 1.4 has t38 pass-through and 1.6 has pass-through and termination, but 1.6 was just release and I would not suggest using it in a production environment unless you can tolerate problem or even outages. If you are planning on using a T1/E1 then send incoming calls to iaxmodem/hylafax or to an ATA/FXS card. Either works very well. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Brendan Martens Sent: Wednesday, October 22, 2008 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] fax / t38 gateway I'm trying to figure out how to handle our fax line when we switch to our asterisk for voice. After a lot of reading and poking about I have concluded, as have many others it would seem, that the best thing to do is either to have a separate pstn fax line or use some sort of internet faxing service rather than try and make faxing work in a way it's not meant to over voip lines. The question I can't seem to find a good answer to is if there is a service/software that would allow a DID to be transferred to them and then they perform the t.38 gateway/conversion functions to which I can connect with asterisk as a t.38 endpoint and originator, or if there is a way that I could host that on my own server? So essentially I am a bit confused that asterisk supports t.38 as an endpoint or originator, but there doesn't seem to be a way to convert to/from analog for interoperating with normal fax machines. I'm sure something exists or the code wouldn't have been written into asterisk... Can someone point me in the right direction? Brendan Martens ___ -- 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
