Re: [asterisk-users] 3way calling / codec problem
Right - I get the error on the console - I just can't tell how many "transcodes" are occuring at any given point in time... On 10/18/06, Thomas Kenyon <[EMAIL PROTECTED]> wrote: Mr. Jones wrote: > Is there some way I can tell? > > On 10/16/06, Thomas Kenyon <[EMAIL PROTECTED]> wrote: >> Mr. Jones wrote: >> > I'm having problems with conference calls (3-way) when I have my codec >> > forced to g729 in sip.conf. >> > >> > I'm using Grandstream 2000s. >> > >> > If enable both g711 and g729 then 3 way calling and transfers work. >> > >> > I'm not sure why this would matter? >> > >> > Here's the error: >> > >> > Oct 13 13:54:45 NOTICE[31184] chan_sip.c: No compatible codecs! >> > >> > Any help is greatly appreciated! >> >> Are you out of licences? From memory when in a console each channel >> needs to be able to be transcoded to SLIN. (where it is mixed and >> transcoded back again). I meant conference (not console). You can show g729 or have a console open with verbosity set (probably to 3) and it should tell you on the console output (usually several times). ___ --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] 3way calling / codec problem
Is there some way I can tell? On 10/16/06, Thomas Kenyon <[EMAIL PROTECTED]> wrote: Mr. Jones wrote: > I'm having problems with conference calls (3-way) when I have my codec > forced to g729 in sip.conf. > > I'm using Grandstream 2000s. > > If enable both g711 and g729 then 3 way calling and transfers work. > > I'm not sure why this would matter? > > Here's the error: > > Oct 13 13:54:45 NOTICE[31184] chan_sip.c: No compatible codecs! > > Any help is greatly appreciated! Are you out of licences? From memory when in a console each channel needs to be able to be transcoded to SLIN. (where it is mixed and transcoded back again). ___ --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] 3way calling / codec problem
I'm having problems with conference calls (3-way) when I have my codec forced to g729 in sip.conf. I'm using Grandstream 2000s. If enable both g711 and g729 then 3 way calling and transfers work. I'm not sure why this would matter? Here's the error: Oct 13 13:54:45 NOTICE[31184] chan_sip.c: No compatible codecs! Any help is greatly 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
Re: [asterisk-users] transfer from VM to Cell Phone
Thanks guys! I was hoping to let them leave a voicemail, then transfer to cell - in case the user doesn't answer at least they get to leave a message. On 10/10/06, Chris Ramsey <[EMAIL PROTECTED]> wrote: I don't think you would need a macro for this. After Asterisk determines that their first extention is unavailable, give the user the choice of trying the cell or just going to VM via a background command. On 10/10/06, Dovid B <[EMAIL PROTECTED]> wrote: > You can create a macro that tells the caller that the user is unavailable. > It then asks them if they want to go to the usersVM or be transfred to thier > cell phone. > > I also created a macro where users can dial an extension and set thier > mobile number. Let me know if you want it. > > > - Original Message - > From: "Mr. Jones" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, October 10, 2006 8:45 PM > Subject: [asterisk-users] transfer from VM to Cell Phone > > > > Hi Folks, > > > > I'm not sure if this is possible, but I'd like to give users the > > option of transfering to an employee's cell phone when they get to > > their greeting. This is a feature that is common on Nortel KSUs. > > > > Is there an "easy" way to do this on a per employee basis? I can see > > how it can be done globally. > > > > TIA > > > > Brian, > > ___ > > --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 > -- -- www.AsteriskBlog.com Your home for easy to learn Asterisk stuff. ___ --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] transfer from VM to Cell Phone
Hi Folks, I'm not sure if this is possible, but I'd like to give users the option of transfering to an employee's cell phone when they get to their greeting. This is a feature that is common on Nortel KSUs. Is there an "easy" way to do this on a per employee basis? I can see how it can be done globally. TIA Brian, ___ --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] Forcing Transcode
Thanks - This worked. I swear I was getting a 503 or something weird before when I did this but it seems to be working now. On 9/28/06, Andres <[EMAIL PROTECTED]> wrote: Mr. Jones wrote: > Hi Folks, > > I'm curious if there's anyway to force Asterisk to transcode for > certain handsets. All you need is to edit your sip.conf and force the codec per user entry: [your entry here] disallow=all allow=g729 > > Specifically we have an inbound SIP origination service which uses g711. > > We're having bandwidth issues with a client and would like to force > Asterisk to transcode to g729 until we can get their T1 in place. > > Any ideas? > ___ > --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 > > -- Andres Technical Support http://www.telesip.net ___ --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] Forcing Transcode
Hi Folks, I'm curious if there's anyway to force Asterisk to transcode for certain handsets. Specifically we have an inbound SIP origination service which uses g711. We're having bandwidth issues with a client and would like to force Asterisk to transcode to g729 until we can get their T1 in place. Any ideas? ___ --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] Got SIP response 415 "Unacceptable Content-Type" back from 192.168.1.209
I'm still getting these errors if anyone has any ideas I'd be truly appreciative. On 9/25/06, Mr. Jones <[EMAIL PROTECTED]> wrote: Could the problem is this: "Content-Type: unknown"? Reliably Transmitting (NAT) to 192.168.1.228:5060: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK4bf724c9;rport From: ;tag=as744e33c0 To: "test guy" ;tag=6583e0d3a15652bd Contact: Call-ID: [EMAIL PROTECTED] CSeq: 109 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: - Content-Type: unknown Subscription-State: active Content-Length: 0 --- asterisk*CLI> <-- SIP read from 192.168.1.228:5060: SIP/2.0 415 Unacceptable Content-Type Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK77316e5c;rport From: ;tag=as744e33c0 To: "test guy" ;tag=6583e0d3a15652bd Call-ID: [EMAIL PROTECTED] CSeq: 108 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.9 Accept: application/sdp, application/simple-message-summary, application/octet-stream, application/pidf+xml, message/sipfrag;version=2.0 Content-Length: 0 On 9/25/06, Anthony Cennami <[EMAIL PROTECTED]> wrote: > Bidirectional SIP trace usually helps in these situations. > > > On 9/25/06, Mr. Jones <[EMAIL PROTECTED]> wrote: > > > > Hi Folks, > > > > Has anyone seen these errors repeatedly in the CLI? > > > > Incoming call: Got SIP response 415 "Unacceptable Content-Type" back > > from 192.168.1.209 > > > > We're using GXP-2000s. > > > > TIA, > > > > Brian > > ___ > > --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 > > > > > > -- > Anthony D Cennami > > ___ > --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] Got SIP response 415 "Unacceptable Content-Type" back from 192.168.1.209
Could the problem is this: "Content-Type: unknown"? Reliably Transmitting (NAT) to 192.168.1.228:5060: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK4bf724c9;rport From: ;tag=as744e33c0 To: "test guy" ;tag=6583e0d3a15652bd Contact: Call-ID: [EMAIL PROTECTED] CSeq: 109 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: - Content-Type: unknown Subscription-State: active Content-Length: 0 --- asterisk*CLI> <-- SIP read from 192.168.1.228:5060: SIP/2.0 415 Unacceptable Content-Type Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK77316e5c;rport From: ;tag=as744e33c0 To: "test guy" ;tag=6583e0d3a15652bd Call-ID: [EMAIL PROTECTED] CSeq: 108 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.9 Accept: application/sdp, application/simple-message-summary, application/octet-stream, application/pidf+xml, message/sipfrag;version=2.0 Content-Length: 0 On 9/25/06, Anthony Cennami <[EMAIL PROTECTED]> wrote: Bidirectional SIP trace usually helps in these situations. On 9/25/06, Mr. Jones <[EMAIL PROTECTED]> wrote: > > Hi Folks, > > Has anyone seen these errors repeatedly in the CLI? > > Incoming call: Got SIP response 415 "Unacceptable Content-Type" back > from 192.168.1.209 > > We're using GXP-2000s. > > TIA, > > Brian > ___ > --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 > -- Anthony D Cennami ___ --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] Got SIP response 415 "Unacceptable Content-Type" back from 192.168.1.209
Hi Folks, Has anyone seen these errors repeatedly in the CLI? Incoming call: Got SIP response 415 "Unacceptable Content-Type" back from 192.168.1.209 We're using GXP-2000s. TIA, Brian ___ --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] codecs/voicemail/DTMF
Hi Eric, I'm confused on where I would put this? I'm also confused on how this would help with external calls (which we want to be g729) vs internal calls to voicemail (which appear to need to be g711)? Thanks a ton! Brian On 9/19/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: Use type=user for inbound and type=peer for outbound. Have different codec settings for each of them. Mr. Jones wrote: > Hi Folks, > > We're trying to roll Asterisk out to production and are having a few > complications. > > Most specifically we have G711 for our inbound origination, but would > prefer G729 for outbound termination, so far so good - it appears that > dtmfmode=auto works in both cases. > > The area I'm having trouble with is, in order to have g729 on the > outbound I have: > > disallow=all > allow=g729 > allow=ulaw > allow=alaw > > In sip.conf at the [general] level. > > When we call voicemail, or the auto attendant internally touchtones > don't work and we get: > > WARNING[8393]: dsp.c:1422 ast_dsp_process: Inband DTMF is not > supported on codec g729. Use RFC2833 > > I'm just guessing, but I thought "auto" was supposed to negotiate the > DTMF mode. Since it appears that the voicemail can't handle RFC2833, > is there some way to force the codec to resort to G711? > > Thanks! > > Brian > ___ > --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] codecs/voicemail/DTMF
Hi Folks, We're trying to roll Asterisk out to production and are having a few complications. Most specifically we have G711 for our inbound origination, but would prefer G729 for outbound termination, so far so good - it appears that dtmfmode=auto works in both cases. The area I'm having trouble with is, in order to have g729 on the outbound I have: disallow=all allow=g729 allow=ulaw allow=alaw In sip.conf at the [general] level. When we call voicemail, or the auto attendant internally touchtones don't work and we get: WARNING[8393]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 I'm just guessing, but I thought "auto" was supposed to negotiate the DTMF mode. Since it appears that the voicemail can't handle RFC2833, is there some way to force the codec to resort to G711? Thanks! Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP 415 messagse
Hi Folks, I'm getting a lot of these messagse now with the Grandstream phones and Asterisk Incoming call: Got SIP response 415 "Unacceptable Content-Type" back from 192.168.1.X I don't think I noticed them when I only had one or two phones hooked up for testing, but I suppose I could have just missed them andthat the frequency isn't very high. Now with 30 devices they are very frequent. Any ideas? ___ --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] Queue timeout problems
Hi Guido - Evidently I needed to add a timeout to the queue itself. Thanks, Brian On 9/3/06, Guido Hecken <[EMAIL PROTECTED]> wrote: > -Ursprüngliche Nachricht- > Von: Mr. Jones [mailto:[EMAIL PROTECTED] > Gesendet: Sonntag, 3. September 2006 06:10 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: Re: [asterisk-users] Queue timeout problems > > Thanks Guido - > > I tried that and still have the same problem. The call never seems to > leave the queue. > > Any other ideas? Hmm, to have a closer look on the problem, one could do the following Activate debugging, error and verbose logging in logger.conf by having a line like this: console => notice,warning,error,debug,verbose Open the cli and do a logger reload set verbose to 5 or even 255 Initiate a call to the queue and watch for errors/informations. Perhaps, define a context named test and put a really simple command in it. Something like this [test] exten => 120,1,Answer() exten => 120,2,Playback(some-sound-file) exten => 120,3,Hangup Change your queue to call this context in the second priority. Also have a closer look on your include commands in the dialplan... Normally an extensions reload on the cli should activate the changes to the dialplan, but with a restart now you should be save. Good luck Guido ___ --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] Queue timeout problems
Thanks Guido - I tried that and still have the same problem. The call never seems to leave the queue. Any other ideas? On 9/2/06, Guido Hecken <[EMAIL PROTECTED]> wrote: > -Ursprüngliche Nachricht- > Von: Mr. Jones [mailto:[EMAIL PROTECTED] > Gesendet: Sonntag, 3. September 2006 01:12 > An: asterisk-users@lists.digium.com > Betreff: [asterisk-users] Queue timeout problems > > Hi Folks, > > I'm trying to use the Queue feature to essentially implement a > multiple call appearance situation for some of our executives. > > Essentially I have a queue defined per executive like: > exten=>9495551212,1, Queue(stever|tTr|||25) > exten=>9495551212,2, Goto(druid-users,1212,1) Give these settings a try: exten=>9495551212,1,Wait(2) exten=>9495551212,2,(Playback(some-announce) ; could be an empty sound file exten=>9495551212,3,Queue(stever|tT|||60) ; try without option r exten=>9495551212,4,Goto(druid-users,1212,1) [stever] strategy=ringall context=druid-default joinempty=yes member=> SIP/1200 member=> SIP/1201 member=> SIP/1212 timeout=15 > > So the user hits the queue ok, but they never fallout to the 2nd > priorty, which has macros for follow-me, and handles the voice mail. What happens to the call instead? Dropped, endless in the queue? > > in queues.conf I have > > [stever] > strategy=ringall > context=druid-default > joinempty=yes > member=> SIP/1200 > member=> SIP/1201 > member=> SIP/1212 > timeoutreset = no Hope, it helps Guido ___ --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] Queue timeout problems
Hi Folks, I'm trying to use the Queue feature to essentially implement a multiple call appearance situation for some of our executives. Essentially I have a queue defined per executive like: exten=>9495551212,1, Queue(stever|tTr|||25) exten=>9495551212,2, Goto(druid-users,1212,1) So the user hits the queue ok, but they never fallout to the 2nd priorty, which has macros for follow-me, and handles the voice mail. in queues.conf I have [stever] strategy=ringall context=druid-default joinempty=yes member=> SIP/1200 member=> SIP/1201 member=> SIP/1212 timeoutreset = no I've tried with timeoutreset set to both no and yes. TIA! ___ --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] Operator Console(s)/Shared Call Appearances
Hi Folks, I'm back on this subject again. What's the best way to have both phones ring simultaneously across say 3 operators + the executive? TIA On 7/24/06, Jerry Jones <[EMAIL PROTECTED]> wrote: Asterisk does not yet support bridged calls You can easily have a button labeled exec 1 ring on her phone at the same time it rings the execs phone, and have one light if he is on the phone Also FOP works great On Jul 23, 2006, at 3:42 PM, Mr. Jones wrote: > Thanks Sebastian - > > You're right - I have limited experience in this area :) > > I think the idea below is workable, except we actually want it to work > in the other direction - sort of. > > Essentially we want the receptionist to screen the calls when she's > available. The executive should have option to answer the phone if its > after hours, or they know the receptionist isn't available (or perhaps > they recognize the caller ID and just want to take the call). > > Can you think of how this might work? I suppose the executive could be > a member of his own queue? > >> What do you think about this idea; >> 1. Call comes in at one of the executive numbers. >> 2. Executive phone starts ringing for a predetermined time. >> 3. The callerid is changed to also reflect the name/number of called >> executive, so that the receptionist knows for who the call was. >> 4. The call is dropped into a queue for the receptionist (queue >> because >> multiple calls to the receptionist at the same time are possible). >> >> >> This setup isn't all that hard, and doesn't require more than 4 sip >> accounts / phones and one queue, with one agent. Furthermore, if your >> company starts to grow, and more receptionists that have to answer >> the >> phone are needed, it's quite easy, all you have to do is add a sip >> account, one agent and add that agent to the existing queue. (About 2 >> minutes...) >> >> -- >> Sebastian Berm >> iPronto Communications >> ___ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: getting SIP to listen on multiple ports
Please disregard this message. Evidently changing the port required a power cycle on the PAP2. On 8/26/06, Mr. Jones <[EMAIL PROTECTED]> wrote: Is it possible to get sip to listen on two ports (say 5060 and 5061)? Maybe its not necessary, but I'm trying to get a PAP2 to work with 2 lines configured behind a Linksys router with NAT. I've noticed the default config in the PAP2 is to use 5060 for line 1 and 5061 for line 2. I'm guessing this is to assist in the handling of SIP through a NAT. If I try using 5060 for both lines I never see a registration for line 2. Any ideas? TIA ___ --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] getting SIP to listen on multiple ports
Is it possible to get sip to listen on two ports (say 5060 and 5061)? Maybe its not necessary, but I'm trying to get a PAP2 to work with 2 lines configured behind a Linksys router with NAT. I've noticed the default config in the PAP2 is to use 5060 for line 1 and 5061 for line 2. I'm guessing this is to assist in the handling of SIP through a NAT. If I try using 5060 for both lines I never see a registration for line 2. Any ideas? TIA ___ --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] SIP header challenge
Thanks Rich - Maybe I'll try the dev mailing list. I'm not that familiar with the protocol level as well. I'm thinking its related to one of those two items (user=phone or the Contact: being blank). I've looked through all the configs and I don't seem to see any way to have the Contact fall back to parsing the SIP To: field. For now I have a macro which does this "ok" but I can't use and of the web interface tools to manage these extensions as I basically have to bypass the extension pass the "extension" in to a 2nd macro for dialing, voicemail, ec. RJ On 8/13/06, Rich Adamson <[EMAIL PROTECTED]> wrote: > > I'm using a wholesale voip origination provider - they don't deal with > end users. As such they have statically defined my Asterisk box on > their end - there's no registration or authentication by my system > with theirs - other than them hardcoding the destination IP of my > server in their system. > > As such I don't have a username to "register" with them. Additionally > they will be originating 100s of DIDs for me at the end of the day, so > it would be horrible to have to register all these in some way. > > Also, I don't have any "register" statements with provider A, and > somehow that seems to work out just fine as well. > > It seems to me like Asterisk doesn't like the format of what's coming > from their system. Perhaps the addition of: ";user=phone" is confusing > Asterisk? > > That's the only obvious difference, that and the Contact: header being > set to "s". > > Any other ideas? Nope, other then if I were trying to troubleshoot the issue, I'd get an ethereal trace of both A and B along with a matching sip debug. By comparing the two trace methods one should be able to rule out at least some issues and narrow the possible root cause. I don't consider myself a sip protocol expert, but I'd have to guess that Olle, Kevin, and few others on the list would be able to help interpret that output if its condensed to some reasonable size, and not summarized by selected copy/paste of portions of the traces that might miss important details. Given the level of detail and amount of analysis time needed to fully understand the issue, posting all of that to the list probably isn't going to get the wanted result. Maybe a private email to one of the more knowledgeable sip folks requesting their assistance might be helpful. If the problem turns out to be something like the ";user=phone" mentioned, I'd have to wonder if the wholesale provider would actually do anything about it though. R. ___ --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] SIP header challenge
Hi Rich, I'm using a wholesale voip origination provider - they don't deal with end users. As such they have statically defined my Asterisk box on their end - there's no registration or authentication by my system with theirs - other than them hardcoding the destination IP of my server in their system. As such I don't have a username to "register" with them. Additionally they will be originating 100s of DIDs for me at the end of the day, so it would be horrible to have to register all these in some way. Also, I don't have any "register" statements with provider A, and somehow that seems to work out just fine as well. It seems to me like Asterisk doesn't like the format of what's coming from their system. Perhaps the addition of: ";user=phone" is confusing Asterisk? That's the only obvious difference, that and the Contact: header being set to "s". Any other ideas? TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP header challenge
This is essentially a follow-up to my previous email on the 404 I was seeing with my DIDs. I think it maybe more involved with the SIP headers I'm receiving from the company providing my origination. Here's what's interesting. I have inbound 800 service and outbound termination from provider A. This is working great. I don't have anything specific configured in sip.conf, and the dialed 800 number matches an entry in extensions.conf - everything works great. With provider B who is providing my nationwide origination I'm having some challenges. I'm having to extract the SIP header and modify it then pass it into a macro to actually route it to extensions. Its not so pretty. In digging into the SIP headers I've noticed the following. The first sample is from Provider "A" and works: To: ;tag=as6519d118 I also noticed the "Contact" header seems to be filled in: Contact: From provider B: To: ;tag=as7154d7ed Contact: So I guess my question is two fold: 1. What sets the "Contact" field - Asterisk, or the sending machine? 2. Would the ;user=phone and the lack of a port number be causing Asterisk not to set the exten properly? TIA ___ --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] Inbound Calls & SIP/2.0 404 Not Found
So it looks like the information is coming through in the SIP header.
Is there anyway to avoid the "register" command - at the end of the
day I may have 100s of DIDs and I don't want to have to set them up by
hand.
Is it possible to "fix" what Asterisk thinks the extension is be "resetting" it?
Something like this:
exten => s,1,Set(EXTEN = ${SIP_HEADER(TO):5:10}))
I don't seem to be having any luck with this command.
Thanks!
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Re: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found
Here you go:
<-- SIP read from 1.2.3.4:5060:
INVITE sip:3125551212;[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK08mh9j107ol1tace2500.1
From: ;tag=1000-0-1617457931
To:
CSeq: 1 INVITE
Contact:
Call-ID: [EMAIL PROTECTED]
P-Asserted-Identity:
Privacy: none
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 228
v=0
o=- 3364317821 3364317821 IN IP4 1.2.3.4
s=-
c=IN IP4 1.2.3.4
t=0 0
m=audio 20042 RTP/AVP 0 8 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (12 headers 11 lines)---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 1.2.3.4 : 5060 (non-NAT)
Found peer 'paetec_inbound'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 1.2.3.4:20042
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for s in lance-test (domain 3125551212)
list_route: hop:
Transmitting (NAT) to 1.2.3.4:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
1.2.3.4:5060;branch=z9hG4bK08mh9j107ol1tace2500.1;received=1.2.3.4
From: ;tag=1000-0-1617457931
To:
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact:
Content-Length: 0
---
-- Executing NoOp("SIP/5060-b7a1aa50", "Exten is: s") in new stack
== Auto fallthrough, channel 'SIP/5060-b7a1aa50' status is 'UNKNOWN'
Reliably Transmitting (NAT) to 1.2.3.4:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
1.2.3.4:5060;branch=z9hG4bK08mh9j107ol1tace2500.1;received=1.2.3.4
From: ;tag=1000-0-1617457931
To: ;tag=as48adf352
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact:
Content-Length: 0
---
asterisk*CLI>
<-- SIP read from 1.2.3.4:5060:
ACK sip:3125551212;[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK08mh9j107ol1tace2500.1
CSeq: 1 ACK
From: ;tag=1000-0-1617457931
To: ;tag=as48adf352
Call-ID: [EMAIL PROTECTED]
Max-Forwards: 69
Content-Length: 0
--- (8 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
On 8/11/06, Nir Simionovich <[EMAIL PROTECTED]> wrote:
Hmmm...
Appears as if the SIP invite request is ill-formed. Can you send the SIP
debug
of the session to the list, so we may examine it?
Nir S
-Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Mr. Jones
Sent: Friday, August 11, 2006 10:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found
Actually it looks like I am getting the number but its coming through weird:
This is what sip debug gives me:
Looking for s in test-context (domain 9495551212)
So clearly I am getting the number, just not sure if its formated ok?
On 8/11/06, Mr. Jones <[EMAIL PROTECTED]> wrote:
> Yeah...
>
>
> I tried the NoOp function someone gave me above and I'll I'm getting is
"s"
>
> I'll go back to the provider
>
> On 8/11/06, C F <[EMAIL PROTECTED]> wrote:
> > s, means that it got an incoming call, but no exten came with it.
> >
> > On 8/11/06, Mr. Jones <[EMAIL PROTECTED]> wrote:
> > > I double checked the context.
> > >
> > > But the "Looking for s" is a bit confusing - not sure what "s" is?
> > >
> > > On 8/11/06, Vadim Berezniker <[EMAIL PROTECTED]> wrote:
> > > > Perhaps the context in sip.conf doesn't match the context in the
dial plan.
> > > >
> > > >
> > > >
> > > > From: [EMAIL PROTECTED] on
behalf of Mr. Jones
> > > > Sent: Fri 8/11/2006 2:34 PM
> > > > To: asterisk-users@lists.digium.com
> > > > Subject: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found
> > > >
> > > >
> > > >
> > > > I'm trying to get inbound DIDs working via SIP.
> > > >
> > > > I have 20 DIDs coming in via a single SIP profile in sip.conf.
> > > >
> > > > I was hoping to have these matched in extensions.conf, so I have
setup
> > > > lines like this:
> > > >
> > > > exten=>949271,1, Goto(mainmenu,s,1)
> > > >
> > >
Re: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found
Ok - now maybe we're getting somewhere. I didn't know I had to register them? This is inbound only and the provider doesn't require that - so do I just makeup a username? I currently have the provider as a SIP peer. On 8/11/06, Rushowr <[EMAIL PROTECTED]> wrote: Uh, what's your Register statement for those SIP DIDs look like? If you don't specify the number after a /, you'll be handed calls for that line, but specifying 's' as the extension. register => user[:secret[:[EMAIL PROTECTED]:port][/extension] I consider that last argument required anymore Sherwood -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, August 11, 2006 3:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found You might want to take a look at 'sip debug' to see what your provider is actually sending you. Its likely they aren't sending you the 9495551212 sting as you are expecting. > Thanks - > > Just to be clear - I just replaced the real digits with - I want > to direct these to specific extensions. So maybe I should have used > or something else? > > I tried this: > > exten=>_9495551212,1, Goto(mainmenu,s,1) > > But still to no avail. > > On 8/11/06, Vadim Berezniker <[EMAIL PROTECTED]> wrote: >> Perhaps the context in sip.conf doesn't match the context in the dial >> plan. >> >> >> >> >> I'm trying to get inbound DIDs working via SIP. >> >> I have 20 DIDs coming in via a single SIP profile in sip.conf. >> >> I was hoping to have these matched in extensions.conf, so I have >> setup lines like this: >> >> exten=>949271,1, Goto(mainmenu,s,1) >> >> Unfortunately these aren't getting matched and I'm getting this error: >> >> Looking for s in druid-default (domain 949271) SIP/2.0 404 Not >> Found >> >> Any hints or tips? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found
I have that
Looking for s in test-context (domain 9495551212)
-- Executing NoOp("SIP/5060-b7a1aa50", "Exten is: s") in new stack
== Auto fallthrough, channel 'SIP/5060-b7a1aa50' status is 'UNKNOWN'
SIP/2.0 603 Declined
I'm just not sure how to use the "domain BLAH" to match an extension.
On 8/11/06, Hermann Wecke <[EMAIL PROTECTED]> wrote:
Mr. Jones wrote:
> I have 20 DIDs, some I want to send to a menu, most directly to an
> extension.
sip debug is (really) your friend. It should give you the [context]
where your DID is being send to and the 404 not found error also.
A particular line to look for: "Looking for ...".
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Re: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found
Actually it looks like I am getting the number but its coming through weird: This is what sip debug gives me: Looking for s in test-context (domain 9495551212) So clearly I am getting the number, just not sure if its formated ok? On 8/11/06, Mr. Jones <[EMAIL PROTECTED]> wrote: Yeah... I tried the NoOp function someone gave me above and I'll I'm getting is "s" I'll go back to the provider On 8/11/06, C F <[EMAIL PROTECTED]> wrote: > s, means that it got an incoming call, but no exten came with it. > > On 8/11/06, Mr. Jones <[EMAIL PROTECTED]> wrote: > > I double checked the context. > > > > But the "Looking for s" is a bit confusing - not sure what "s" is? > > > > On 8/11/06, Vadim Berezniker <[EMAIL PROTECTED]> wrote: > > > Perhaps the context in sip.conf doesn't match the context in the dial plan. > > > > > > > > > > > > From: [EMAIL PROTECTED] on behalf of Mr. Jones > > > Sent: Fri 8/11/2006 2:34 PM > > > To: asterisk-users@lists.digium.com > > > Subject: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found > > > > > > > > > > > > I'm trying to get inbound DIDs working via SIP. > > > > > > I have 20 DIDs coming in via a single SIP profile in sip.conf. > > > > > > I was hoping to have these matched in extensions.conf, so I have setup > > > lines like this: > > > > > > exten=>949271,1, Goto(mainmenu,s,1) > > > > > > Unfortunately these aren't getting matched and I'm getting this error: > > > > > > Looking for s in druid-default (domain 949271) > > > SIP/2.0 404 Not Found > > > > > > Any hints or tips? > > > > > > TIA > > > ___ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > ___ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --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] Inbound Calls & SIP/2.0 404 Not Found
Yeah... I tried the NoOp function someone gave me above and I'll I'm getting is "s" I'll go back to the provider On 8/11/06, C F <[EMAIL PROTECTED]> wrote: s, means that it got an incoming call, but no exten came with it. On 8/11/06, Mr. Jones <[EMAIL PROTECTED]> wrote: > I double checked the context. > > But the "Looking for s" is a bit confusing - not sure what "s" is? > > On 8/11/06, Vadim Berezniker <[EMAIL PROTECTED]> wrote: > > Perhaps the context in sip.conf doesn't match the context in the dial plan. > > > > > > > > From: [EMAIL PROTECTED] on behalf of Mr. Jones > > Sent: Fri 8/11/2006 2:34 PM > > To: asterisk-users@lists.digium.com > > Subject: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found > > > > > > > > I'm trying to get inbound DIDs working via SIP. > > > > I have 20 DIDs coming in via a single SIP profile in sip.conf. > > > > I was hoping to have these matched in extensions.conf, so I have setup > > lines like this: > > > > exten=>949271,1, Goto(mainmenu,s,1) > > > > Unfortunately these aren't getting matched and I'm getting this error: > > > > Looking for s in druid-default (domain 949271) > > SIP/2.0 404 Not Found > > > > Any hints or tips? > > > > TIA > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Inbound Calls & SIP/2.0 404 Not Found
Thanks Kevin - I realized afterwards that the was a bad example. It should be a specific number, I was just masking it. I have 20 DIDs, some I want to send to a menu, most directly to an extension. I've tried: exten=>9492711234,1, Macro(druiexten,3711,SIP/3711) and: exten=>_9492711234,1, Macro(druiexten,3711,SIP/3711) Thanks On 8/11/06, Kevin Smith <[EMAIL PROTECTED]> wrote: If I am following you right, for extension matching you need to have a "_" in front of the number. So your example should be like this: exten => _949927,1,Goto(mainmenu,s,1) Also I don't know if you did this on purpose or not but N will only match for numbers 2-9, if you want 0-9 you will want to use an X. Otherwise without the "_" in front of the number it will not extension pattern match. There are other pattern matching characters too, but you can read them here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns Kevin Mr. Jones wrote: > I'm trying to get inbound DIDs working via SIP. > > I have 20 DIDs coming in via a single SIP profile in sip.conf. > > I was hoping to have these matched in extensions.conf, so I have setup > lines like this: > > exten=>949271,1, Goto(mainmenu,s,1) > > Unfortunately these aren't getting matched and I'm getting this error: > > Looking for s in druid-default (domain 949271) > SIP/2.0 404 Not Found > > Any hints or tips? > > TIA > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found
I double checked the context. But the "Looking for s" is a bit confusing - not sure what "s" is? On 8/11/06, Vadim Berezniker <[EMAIL PROTECTED]> wrote: Perhaps the context in sip.conf doesn't match the context in the dial plan. From: [EMAIL PROTECTED] on behalf of Mr. Jones Sent: Fri 8/11/2006 2:34 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found I'm trying to get inbound DIDs working via SIP. I have 20 DIDs coming in via a single SIP profile in sip.conf. I was hoping to have these matched in extensions.conf, so I have setup lines like this: exten=>949271,1, Goto(mainmenu,s,1) Unfortunately these aren't getting matched and I'm getting this error: Looking for s in druid-default (domain 949271) SIP/2.0 404 Not Found Any hints or tips? TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found
Thanks - Just to be clear - I just replaced the real digits with - I want to direct these to specific extensions. So maybe I should have used or something else? I tried this: exten=>_9495551212,1, Goto(mainmenu,s,1) But still to no avail. On 8/11/06, Vadim Berezniker <[EMAIL PROTECTED]> wrote: Perhaps the context in sip.conf doesn't match the context in the dial plan. From: [EMAIL PROTECTED] on behalf of Mr. Jones Sent: Fri 8/11/2006 2:34 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found I'm trying to get inbound DIDs working via SIP. I have 20 DIDs coming in via a single SIP profile in sip.conf. I was hoping to have these matched in extensions.conf, so I have setup lines like this: exten=>949271,1, Goto(mainmenu,s,1) Unfortunately these aren't getting matched and I'm getting this error: Looking for s in druid-default (domain 949271) SIP/2.0 404 Not Found Any hints or tips? TIA ___ --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] Inbound Calls & SIP/2.0 404 Not Found
I'm trying to get inbound DIDs working via SIP. I have 20 DIDs coming in via a single SIP profile in sip.conf. I was hoping to have these matched in extensions.conf, so I have setup lines like this: exten=>949271,1, Goto(mainmenu,s,1) Unfortunately these aren't getting matched and I'm getting this error: Looking for s in druid-default (domain 949271) SIP/2.0 404 Not Found Any hints or tips? TIA ___ --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] Warning - Voiplink.com doesn't deliver - stuck in a hole
I have had the same experience with a Grandstream order from them - 7 days and no product. They even told me it was shipping Monday, but couldn't produce a tracking number on Tuesday. Pretty lame. On 8/9/06, Tom <[EMAIL PROTECTED]> wrote: Is anyone else having problems with them? Order placed online 13 days ago. voiplink.com charged my cc for the product 11 days ago. They can't seem to ship Linksys spa-942 that they claim to have in stock. Order is still pending on their web site. Calls to them confirm no shipment but also get no results. Anyone know of a good supplier for the spa-942? ___ --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] Re: Operator Console(s)/Shared Call Appearances
Yes this is what I want. I guess the question is what is the best way to do it? Use a Queue? or something else? On 25 Jul 2006 13:25:45 +0200, Benny Amorsen <[EMAIL PROTECTED]> wrote: > "J" == Jones <[EMAIL PROTECTED]> writes: J> Hi Folks, We're migrating from a conventional KSU/PBX to Asterisk J> and I'm trying to determine the best way to allow our receptionist J> to answer certain executives telephone lines. J> It seems there are probably two routes, but I'm not sure of the J> limitations of each. You could make both the executive and the receptionist phones ring, perhaps with a very low ring tone for the executives. Then the receptionist will take the call whenever possible. If the call needs to go through to the executive, the receptionist can do a direct call just by pressing a button, and a different (perhaps louder) ring tone can play. /Benny ___ --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] Operator Console(s)/Shared Call Appearances
Thanks Sebastian - You're right - I have limited experience in this area :) I think the idea below is workable, except we actually want it to work in the other direction - sort of. Essentially we want the receptionist to screen the calls when she's available. The executive should have option to answer the phone if its after hours, or they know the receptionist isn't available (or perhaps they recognize the caller ID and just want to take the call). Can you think of how this might work? I suppose the executive could be a member of his own queue? What do you think about this idea; 1. Call comes in at one of the executive numbers. 2. Executive phone starts ringing for a predetermined time. 3. The callerid is changed to also reflect the name/number of called executive, so that the receptionist knows for who the call was. 4. The call is dropped into a queue for the receptionist (queue because multiple calls to the receptionist at the same time are possible). This setup isn't all that hard, and doesn't require more than 4 sip accounts / phones and one queue, with one agent. Furthermore, if your company starts to grow, and more receptionists that have to answer the phone are needed, it's quite easy, all you have to do is add a sip account, one agent and add that agent to the existing queue. (About 2 minutes...) -- Sebastian Berm iPronto Communications ___ --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] Operator Console(s)/Shared Call Appearances
Hi Folks, We're migrating from a conventional KSU/PBX to Asterisk and I'm trying to determine the best way to allow our receptionist to answer certain executives telephone lines. It seems there are probably two routes, but I'm not sure of the limitations of each. 1. Shared call appearances. This would seem to be the most similar to what we currently have where we have stations/DNs for 3 executives on 3 assistants phones. Of course with the existing system we have lots of programmable buttons. We're leaning towards the SPA-942s, so I'd be interested to know how this might work (do we need the 4 line license, and are we limited to 4 call appearances)? 2. Some form of PC application such as the one at Asternic.org, or something else. This would seem to have the most flexibility, but may require the operator to pay too much attention to the window, unless there's some audible notification. A couple of other alternatives maybe to create a queue, or possibly go with a "side car" type device. I'm open to any and all input. Best, Mr. Jones. ___ --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] DTMF and DISA
ok - so I think IPKall uses g.711. Shouldn't one of the below options work? auto, rfc, or inband? On 6/5/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: DTMF problems happen at the point where the PSTN call is converted to VoIP. EXCEPT where you are using inband DTMF and ulaw or alaw codec. inband DTMF does not work with any other codec. Mr. Jones wrote: > Hi Folks, > > I'm trying to test out Asterisk overall. > > I'm having some problems with DTMF. Currently I'm playing with DISA, > but I'm worried this will happen when I get to implementing AAs etc. > > I have a free SIP trunk from IPKall that I'm trying to make work. > > I'm able to receive calls, and I've now setup and extension with DISA > and a password. > > I connect ok from the PTSN, get the dialtone, and enter the password. > > In the CLI I'm getting duplicate/extra/incorrect digits. > > I've tried dtmfmode=auto, dtmfmode=inband, and dtmfmode=rfc2833 all > with similar results. > > For testing I set the password to 67891 here's what I'm getting (small > sample size, but its pretty random): > > app_disa.c:292 disa_exec: DISA on chan SIP/66.54.140.46-09078d00 got > bad password 677891 > app_disa.c:292 disa_exec: DISA on chan SIP/66.54.140.46-09078cc8 got > bad password 6709 > app_disa.c:292 disa_exec: DISA on chan SIP/66.54.140.46-09078cc8 got > bad password 677891 > > Any hints or tips are appreciated. > > B > ___ > --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 > -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] DTMF and DISA
Hi Folks, I'm trying to test out Asterisk overall. I'm having some problems with DTMF. Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the PTSN, get the dialtone, and enter the password. In the CLI I'm getting duplicate/extra/incorrect digits. I've tried dtmfmode=auto, dtmfmode=inband, and dtmfmode=rfc2833 all with similar results. For testing I set the password to 67891 here's what I'm getting (small sample size, but its pretty random): app_disa.c:292 disa_exec: DISA on chan SIP/66.54.140.46-09078d00 got bad password 677891 app_disa.c:292 disa_exec: DISA on chan SIP/66.54.140.46-09078cc8 got bad password 6709 app_disa.c:292 disa_exec: DISA on chan SIP/66.54.140.46-09078cc8 got bad password 677891 Any hints or tips are appreciated. B ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + PRI Card -> Nortel BCM
Excellent. - So I can basically make a crossover cable to my Nortel, and pass calls to the old phones from the PTSN (via my VOIP originator ) in to it? I guess I'm off to look for sample configs. Thx Brian On 6/3/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: - Mr. Jones <[EMAIL PROTECTED]> wrote: > I imagine some of the PRI cards can "emulate" a switch? Asterisk (and Zaptel) handles all call signaling not the cards. What that means to you is that any Asterisk-supporting T1/E1 card can operate in PRI mode and act as the network end as well as acting as CPE. Many people use Asterisk in exactly the fashion you are describing. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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] Asterisk + PRI Card -> Nortel BCM
Has anyone fed a Nortel BCM from Asterisk? I'm interested in switching our company over, but don't want to replace all the handsets in one fell swoop. I imagine some of the PRI cards can "emulate" a switch? I'd still like to pass CallerID into the Nortel, etc but all the external traffic would be VOIP, not TDM. ___ --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] Re: PAP2-NA Authentication Issues
Please disregard - sometimes I think posting these emails is the key to solving all my problems ;) Evidently I wasn't reloading the files (I thought I was). Brian On 5/31/06, Mr. Jones <[EMAIL PROTECTED]> wrote: Hello Folks, I'm an Asterisk newbie, that being said I have managed to get an SPA941 working with 1.2.8. I've got some issues (like getting the voicemail button to work as it should, and making the message indicator light work) but overall I'm pretty happy. I'm now trying to get a PAP2-NA to work. I reset it, have admin access, updated the firmware, and have the same SIP settings as best I can tell as I do on the 941. I configured a new extension (I'm using Druid), and I've checked sip.conf and it appears valid. I'm getting: May 31 12:04:01 NOTICE[2987]: chan_sip.c:11043 handle_request_register: Registration from '2345 ' failed for '192.168.1.69' - Username/auth name mismatch And I've reset the password 3 or 4 times. So I'm a bit at a loss. Any hints/tips are greatly appreciated. TIA, Brian ___ --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] PAP2-NA Authentication Issues
Hello Folks, I'm an Asterisk newbie, that being said I have managed to get an SPA941 working with 1.2.8. I've got some issues (like getting the voicemail button to work as it should, and making the message indicator light work) but overall I'm pretty happy. I'm now trying to get a PAP2-NA to work. I reset it, have admin access, updated the firmware, and have the same SIP settings as best I can tell as I do on the 941. I configured a new extension (I'm using Druid), and I've checked sip.conf and it appears valid. I'm getting: May 31 12:04:01 NOTICE[2987]: chan_sip.c:11043 handle_request_register: Registration from '2345 ' failed for '192.168.1.69' - Username/auth name mismatch And I've reset the password 3 or 4 times. So I'm a bit at a loss. Any hints/tips are greatly appreciated. TIA, Brian ___ --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
