Re: [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlow bandwidth?
I have indeed already done so - I use G729 quite a bit since I travel alot in adverse conditions. Interesting thing is, I can get less choppy audio sometimes from my Vonage device using (what I suspect to be) Ulaw, while either ulaw or G729 will still give choppy response at that moment from my Linksys Paul - Original Message - From: Marcel van Kaam, Fonetica [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 12:28 AM Subject: RE: [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlow bandwidth? You can set, in the linksys, the codec G729 for your line. In the Linksys also set only to use that codec. This can be done at the admin page of the line you use in the linksys. Also do that in the asterisk for your device. First buy the license from Digium. Then you will use less bandwidth and have a better sound upstream. Marcel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: woensdag 29 juni 2005 1:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceandlow bandwidth? Hmm... Except that if I bring my Vonage ATA for my Vonage line with me to the same hotel, I can get reliable connectivity. Assuming the hotel isn't helping me on the QOS front, and the Hotel's connectivity is the last word, then my Vonage ATA should be choppy, as well, no? This is what leads me to think I can do some tweaking later, Paul - Original Message - From: Greg Oliver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 2:17 PM Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceand low bandwidth? Nothing you can do on this one.. Without the provider accepting your QoS settings, you are at their mercy. And yes, you are correct, most multi-tenant dwellings use xDSL for their connectivity due to it's price, and the upstream is usually less bandwidth than the downstream.. -Greg On Tue, 2005-06-28 at 13:00 -0600, Paul Fielding wrote: So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me a phone at my hotel rooms, etc. During the day or late at night the thing works great - best ATA I've ever used. However, in the mid-evening (when many business travellers are at the hotel room doing work), the outgoing audio channel gets so choppy that the person on the other end can't make me out clearly. Interestingly, I can usually hear them just fine - I attribute that to larger incoming bandwidth than outgoing on the hotel's part. This device has a *lot* of settings that one can tweak. Anyone have any suggestions on tuning this thing (or tuning Asterisk or both) to improve the SIP performance of the audio from the Linksys to the server to try to reduce choppiness? I note that Vonage, who also uses these devices, seems to have got it down - it doesn't seem to matter where I use my Vonage Linksys device, I can get pretty reasonable performance. So I figure I should be able to do similar tweaks to mine... *shrug* regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DID in Western Canada
I tried a Calgary DID with Link2Voip, but they never did get it working correctly. My primary complaint with their customer service is that it was basically non-existant. It took 2 weeks before a service guy even responded to my problem, we fired a few emails back and forth, and then I never, ever, heard from them again. I ended up just giving up and cancelling my DID. It wasn't worth the hassle Paul - Original Message - From: Nelson Loyola [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, June 27, 2005 10:11 AM Subject: [Asterisk-Users] DID in Western Canada Hello, I'm having trouble getting finding a company that provides DID in Western Canada. More specifically in Edmonton, Alberta. We have tried getting in contact with Link2Voip and Calgary Telecom but neither seems to be answering their phones or email. I would appreciate it if anyone can point me in the right direction. Thank you, Nelson __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linksys WRT54GP2-NA settings for performance and low bandwidth?
So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me a phone at my hotel rooms, etc. During the day or late at night the thing works great - best ATA I've ever used. However, in the mid-evening (when many business travellers are at the hotel room doing work), the outgoing audio channel gets so choppy that the person on the other end can't make me out clearly. Interestingly, I can usually hear them just fine - I attribute that to larger incoming bandwidth than outgoing on the hotel's part. This device has a *lot* of settings that one can tweak. Anyone have any suggestions on tuning this thing (or tuning Asterisk or both) to improve the SIP performance of the audio from the Linksys to the server to try to reduce choppiness? I note that Vonage, who also uses these devices, seems to have got it down - it doesn't seem to matter where I use my Vonage Linksys device, I can get pretty reasonable performance. So I figure I should be able to do similar tweaks to mine... *shrug* regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceand low bandwidth?
Hmm... Except that if I bring my Vonage ATA for my Vonage line with me to the same hotel, I can get reliable connectivity. Assuming the hotel isn't helping me on the QOS front, and the Hotel's connectivity is the last word, then my Vonage ATA should be choppy, as well, no? This is what leads me to think I can do some tweaking later, Paul - Original Message - From: Greg Oliver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 2:17 PM Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceand low bandwidth? Nothing you can do on this one.. Without the provider accepting your QoS settings, you are at their mercy. And yes, you are correct, most multi-tenant dwellings use xDSL for their connectivity due to it's price, and the upstream is usually less bandwidth than the downstream.. -Greg On Tue, 2005-06-28 at 13:00 -0600, Paul Fielding wrote: So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me a phone at my hotel rooms, etc. During the day or late at night the thing works great - best ATA I've ever used. However, in the mid-evening (when many business travellers are at the hotel room doing work), the outgoing audio channel gets so choppy that the person on the other end can't make me out clearly. Interestingly, I can usually hear them just fine - I attribute that to larger incoming bandwidth than outgoing on the hotel's part. This device has a *lot* of settings that one can tweak. Anyone have any suggestions on tuning this thing (or tuning Asterisk or both) to improve the SIP performance of the audio from the Linksys to the server to try to reduce choppiness? I note that Vonage, who also uses these devices, seems to have got it down - it doesn't seem to matter where I use my Vonage Linksys device, I can get pretty reasonable performance. So I figure I should be able to do similar tweaks to mine... *shrug* regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] livevoip
I've found the quality to be consistently good - I've tried several other carriers and LiveVOIP is the best I've tried. Their support is indeed a bit blunt, but they get the job done well and quite timely. Paul - Original Message - From: JD Austin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 09, 2005 6:48 PM Subject: [Asterisk-Users] livevoip Anyone use livevoip? opinions? -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVOIP troubleshooting
I think it's a server/connection issue with the LiveVoip server. I'm connected to their Winnipeg server and I get pretty much perfect calling, all the time. A buddy of mine recently got setup on the Vancouver server and is also experiencing choppy audio. He's in the process of asking if he can get moved to the Winnipeg server. We'll see what happens Paul - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 02, 2005 7:21 PM Subject: [Asterisk-Users] LiveVOIP troubleshooting Hi everyone, I need some ideas to troubleshoot this issue: I recently got an 800 numbers from LiveVOIP and it works but on most calls the caller gets hears choppy audio (one drop out per 10 seconds or so). I know this isn't LiveVOIP's support forum but I'm sure some here use their 800 service and I'm interested in their feedback and ideas. And don't get me wrong, LiveVOIP's support has been quite good -- cooperative, fast response, action taken as requested -- but I do not want to try their patience. At this point I am not blaming them for this issue either. Here's the summary: * I'm connected via IAX2 to * The server is in a datacenter with plenty of bandwidth. * Using ulaw with standard 20 ms frames. * I hear the caller perfectly fine, caller hears choppy audio. * tcpdump shows incoming and outgoing packets right on time, every 20 ms in each direction. * I'm not using trunking for now. * Pings to LiveVOIP are about 35 ms. * iax2 show channels shows 1 ms jitter, 42 ms lag. * Drop outs occur on IVR (or audio generated on the server itself) or during normal conversation with a SIP client (ATA or phone) connected to the server remotely. Connection between server and phones is well tested and working fine. I have asked LiveVOIP to switch me from their Vancouver node to their New York node, which reduced ping times from 50 ms to 35 ms. Less chops but still not perfect. Note that the same server is already connected to several Broadvoice accounts, which work flawlessly. Anyway, if anyone has some ideas of what I can try, please let me know. I do not want to keep trying all their nodes to find one that works for me. I do not necessarily want to use a different codec either since I have the bandwidth and I may be receiving faxes, so I need ulaw. Thanks and sorry for the long-ish post. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: LiveVOIP
- Original Message - From: [EMAIL PROTECTED] The fact of the matter is that LiveVoIP has no customer service. They don't care about small users or asterisk users. The few times I've had issues, I've sent off an email and gotten a response within 2 hours. This includes 3am on a Saturday night on a long weekend. The responses may be a bit short and blunt, but they've gotten the job done and quickly. That's much better than I can say for Link2Voip. They activated a DID for me 2 weeks ago and it has yet to work. I've sent multiple emails to them and not so much as a peep in response. No phone number or physical address on their website, only an email address. If I don't hear from them by the end of this week I'm calling my credit card company to cancel the charge - I'm not going to pay for a service they won't provide and won't fix and won't even contact me about... I'll stick with LiveVoip, any day Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO ATA?
- Original Message - The Grandstream HandyTone 488 has an FXO port. I've never used it though. I could be wrong, but I seem to remember reading up on the HandyTone and deciding that it doesn't really act like a true FXO, as in calls come in and go straight to Asterisk like an FXO, and calls can dial out like a true FXO. I think it operated more like an 'ability to dial in number' and as a pass-through in case of power outage. Someone correct me if I'm wrong on that Paul Cheers, Jon. On Thursday 05 May 2005 07:07 am, Chris Mason (Lists) wrote: Is the Sipura 3000 the only way to interface a remote pstn line and connect incoming calls to Asterisk? I have a location connected by network that has a phone line, when the room is occupied I want the line ti ring there as normal, but when the employee is travelling I want the line to be conencted to a ATA that then feeds it as an incoming pstn line to the pbx located at my office so it can follow her. It sounds like the Sipura 3000 would be perfect, what else would do it? Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] livevoip callerid
I wondered that as well, I've tried it with and without the dashes - same
result... :( Number actually works, though, just not name...
regards,
Paul
- Original Message -
From: MF Hulber [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 4:35 AM
Subject: Re: [Asterisk-Users] livevoip callerid
I don't think it's correct to put dashes in the CIDNum.
MARK.
Paul Fielding wrote:
Hmmm... I still can't get name, though number works. Perhaps I'm missing
something?
context livevoip in iax.conf that hooks me to livevoip
dial 9 in front of long distance number to dial livevoip instead of
regular LD.
snip
LIVEVOIP=IAX2/username:[EMAIL PROTECTED]
snip
exten = _91NXXNXX,1,SetCIDNum(403-666-|a)
exten = _91NXXNXX,2,SetCIDName(Satan Lives|a)
exten = _91NXXNXX,3,Noop(Caller Name: ${CALLERIDNAME}, Number:
${CALLERIDNUM})
exten = _91NXXNXX,4,Dial(${LIVEVOIP}/${EXTEN:1})
regards,
Paul
- Original Message - From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 05, 2005 6:31 AM
Subject: Re: [Asterisk-Users] livevoip callerid
I'll be damned... I changed my format to match yours, and both
the SetCIDNum and SetCIDName work just fine. I could never get
the name to work properly prior to your post. Thanks!
I am able to set name and number with Livevoip. Make sure your
variables are actually being set.
exten = s,1,SetCIDNum(xx|a)
exten = s,n,SetCIDName(first last|a)
exten = s,n,Noop(Caller Name: ${CALLERIDNAME}, Number:
${CALLERIDNUM})
MARK.
Cameron Schaus wrote:
Is there any way I can send callerId information to livevoip? I have
added the following to my extensions.conf, but when I place calls
through livevoip, no callerId information is sent to the called party.
SWC_CALLERID=14031234567
SWC_CALLERNAME=foo
exten = _1NXXNXX,1,SetCallerID(${SWC_CALLERID})
exten = _1NXXNXX,2,SetCIDName(${SWC_CALLERNAME})
exten = _1NXXNXX,3,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Thanks,
Cam
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Re: licensing *sigh* (was Re: [Asterisk-Users] US$200 bounty for *paging feature)
- Original Message - From: snacktime [EMAIL PROTECTED] At $200 someone might be willing to do the work if they know it's going to be open source, but if it's a work for hire, $200 is extremely paltry. I'm with you on that one. $200 might be an acceptable bounty to give someone a bit of added incentive to contribute something to the community, but if the code is closed source and owned by the purchaser, than $200 won't even buy a day's worth of real coding. If you want to own it, you don't put out a bounty, you're hiring a programmer, and paying appropriately... regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy
Ok, Here's my ztdummy question. Forgive my ignorance. Everything I read about ztdummy and zaptel cards describes them as being required 'for timing'. But what exactly does this imply? Eg. I have two separate boxes where I did the following: - installed Linux (debian Woody) - compiled a 2.4 kernel - added a few other prereq packages needed to allow Asterisk to compile - compiled and configured Asterisk At this point Asterisk works like a hot darn, no problem, for everything I try to do. No Zaptel card. No ztdummy. So what does ztdummy buy me? regards, Paul - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 13, 2005 5:58 PM Subject: Re: [Asterisk-Users] ztdummy you can get rid of ztdummy. --- Brian Leyton [EMAIL PROTECTED] wrote: I installed a couple of Asterisk test machines, and have been successful in getting them talking to one another, but I have question. After installation, I put an x100p clone in one of the machines. From what I understand, I no longer need ztdummy on that machine, but I'm wondering if it hurts anything. If it's better to remove it, where do I go to get rid of it (I'm running [EMAIL PROTECTED] - which uses CentOS, a Redhat variant)? It looks like it's doing something - have a look at the /proc/interrupts: [EMAIL PROTECTED] root]# more /proc/interrupts CPU0 0:1770360 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 9: 17667040 XT-PIC wcfxo 11: 17694525 XT-PIC ztdummy, usb-uhci, eth0 12: 19 XT-PIC PS/2 Mouse 14: 13676 XT-PIC ide0 15: 12 XT-PIC ide1 NMI: 0 ERR: 0 Brian Leyton IT Manager Commercial Petroleum Equipment ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Liveviop problem
I've actually had sales question oriented calls answered at 2am on a sunday morning, 45 minutes I sent the email (I wasn't expecting a response until monday). Technical email responses at wierd hours as well. No complaints here. I'd send the email again in case it got lost in the shuffle. Having done support myself in the past I can vouch that with the volume of emails support channels get it is always possible for one to occassionally go awal Paul - Original Message - From: Wiley Siler [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 06, 2005 3:14 PM Subject: RE: [Asterisk-Users] Liveviop problem I have had far better luck than that too. More like a hour for me but that is not too bad. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, April 06, 2005 3:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Liveviop problem I'm just curious if someone had/has a problem with livevoip. When I try to make an outgoing call, I receive: -- Called username:secret@217.160.244.186/x037378896 Apr 2 16:47:21 WARNING[10153]: chan_iax2.c:5546 socket_read: Call rejected by 217.160.244.186: No authority found The username,secret and first 5 digits of the phone is modified in this log. I tried to call Livevoip, they said send us an e-mail and I did, but no response whatsoever for about a week now. Been working like a charm here. Just got off the phone. The few times I've had to contact support via email, I had a response with 15 minutes. I might have caught them on a rather slow day though. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] livevoip callerid
Hmmm... I still can't get name, though number works. Perhaps I'm missing
something?
context livevoip in iax.conf that hooks me to livevoip
dial 9 in front of long distance number to dial livevoip instead of regular
LD.
snip
LIVEVOIP=IAX2/username:[EMAIL PROTECTED]
snip
exten = _91NXXNXX,1,SetCIDNum(403-666-|a)
exten = _91NXXNXX,2,SetCIDName(Satan Lives|a)
exten = _91NXXNXX,3,Noop(Caller Name: ${CALLERIDNAME}, Number:
${CALLERIDNUM})
exten = _91NXXNXX,4,Dial(${LIVEVOIP}/${EXTEN:1})
regards,
Paul
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 05, 2005 6:31 AM
Subject: Re: [Asterisk-Users] livevoip callerid
I'll be damned... I changed my format to match yours, and both
the SetCIDNum and SetCIDName work just fine. I could never get
the name to work properly prior to your post. Thanks!
I am able to set name and number with Livevoip. Make sure your
variables are actually being set.
exten = s,1,SetCIDNum(xx|a)
exten = s,n,SetCIDName(first last|a)
exten = s,n,Noop(Caller Name: ${CALLERIDNAME}, Number:
${CALLERIDNUM})
MARK.
Cameron Schaus wrote:
Is there any way I can send callerId information to livevoip? I have
added the following to my extensions.conf, but when I place calls
through livevoip, no callerId information is sent to the called party.
SWC_CALLERID=14031234567
SWC_CALLERNAME=foo
exten = _1NXXNXX,1,SetCallerID(${SWC_CALLERID})
exten = _1NXXNXX,2,SetCIDName(${SWC_CALLERNAME})
exten = _1NXXNXX,3,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Thanks,
Cam
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Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate
Basically, I'm forwarding the standard Asterisk ports: tcp 5060 udp 5060 udp 4569 udp 5036 tcp 5038 udp 5038 udp 1:2 I'm not sure that i needed both tcp and udp on the mgmt port 5038, but what the heck. :) In sip.conf: externip = xx.xx.xx.xx localnet=192.168.1.0 In the sip client contexts they *all* have: nat=yes canreinvite=no This is so that they can be hopped both in and out of NATs without reconfiging. No special ports being forwarded for the clients. They seem to work behind whatever NATs we throw at them without difficulties... later, Paul - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 5:28 AM Subject: RE: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate Thank you for your story Paul, nice work with the dialplans! I have one question, so you say that for server 2, asterisk is behind nat and you have sip clients inside and outside the nat. Which ports are you forwarding to asterisk from your firewall and in the case of sip clients outside nat, did you have to open certain ports for each client or all clients use the same? For inside clients it should be a charm! Very nice job Paul, intercity dialing and everything well connected... That was a good story.. Thx for sharing. Anton -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: Martes, 29 de Marzo de 2005 12:52 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate - Original Message - From: Anton Krall [EMAIL PROTECTED] would like to hear some actual setups and how people are solving the nat issue within scenarios like: Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones I've been playing with this with my friends for awhile now. We've got four different Asterisk servers set up in four different cities: 1. 2 nics - one on internal network, other on external network. TDM400 card with 2 FXO and 1 FXS, 2 different analog lines, and a LiveVoip IAX2 dialout. Various SIP phones connected, both from within the internal network and out on the internet from behind other NATs. 2. 1 nic - behind NAT (ports forwarded). X100p with 1 analog line. Various SIP phones, internal network and from behind other NATs. 3 4. Like #2 but no X100p. All four servers are connected via IAX2 - in all cases we can dial extensions for each other's systems and the call gets dumped to the correct server. Also between server 1 2 we have local inter-city dialing working (if you dial an outside number that is local to the other city and don't put a 1 in front of the number it dumps to the other server and dials out). NAT hasn't proven to be a problem for us - the only thing we can't do as a result of all the SIP clients being natted is Reinvites - this just means that all conversation *must* go through the server as opposed to direct client-client transfer. Servers that are behind nats have the correct IP settings set in SIP.CONF. As long as I set the STUN server on the sip clients to a good working STUN server everything works like a hot damn. Nothing special regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on a dialup connection?
I've actually used xten lite on a mac using GSM codec on a dialup connection. Worked like a hot damn. I was quite surprised, actually. Paul - Original Message - From: cmisip [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, March 27, 2005 7:03 PM Subject: [Asterisk-Users] Asterisk on a dialup connection? How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required unless I can use the onbord G723.1 codecs on the quicknet cards. Ohphone allows this through h323 I think but I want an asterisk solution. If not a fullblown asterisk install on my brothers machine, maybe set it up as a h323 client to mine. I am currently working on setting up one of my lan machines with ohphone to connect to my asterisk box to call FWD and such. Is this possible? Somehow asterisk must translate the codecs from whatever SIP uses to whatever ohphone uses ( I will force it to low bandwitdh g723.1). I am hoping this will work and that the Vonage interconnect will be up soon as this will be a cheap way for them to contact my sister as well. I am still an asterisk newbie so pardon me if the questions seem newbie-ish. Has anybody gone down this path? I hate to have to reinvent the wheel. Anybody have any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate
- Original Message - From: Anton Krall [EMAIL PROTECTED] would like to hear some actual setups and how people are solving the nat issue within scenarios like: Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones I've been playing with this with my friends for awhile now. We've got four different Asterisk servers set up in four different cities: 1. 2 nics - one on internal network, other on external network. TDM400 card with 2 FXO and 1 FXS, 2 different analog lines, and a LiveVoip IAX2 dialout. Various SIP phones connected, both from within the internal network and out on the internet from behind other NATs. 2. 1 nic - behind NAT (ports forwarded). X100p with 1 analog line. Various SIP phones, internal network and from behind other NATs. 3 4. Like #2 but no X100p. All four servers are connected via IAX2 - in all cases we can dial extensions for each other's systems and the call gets dumped to the correct server. Also between server 1 2 we have local inter-city dialing working (if you dial an outside number that is local to the other city and don't put a 1 in front of the number it dumps to the other server and dials out). NAT hasn't proven to be a problem for us - the only thing we can't do as a result of all the SIP clients being natted is Reinvites - this just means that all conversation *must* go through the server as opposed to direct client-client transfer. Servers that are behind nats have the correct IP settings set in SIP.CONF. As long as I set the STUN server on the sip clients to a good working STUN server everything works like a hot damn. Nothing special regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compare with Skype
I use Asterisk because I want the flexibility. My mom uses Skype because it just works. Hey, my Mom can configure Skype. I'll give $100 to the first person that creates a SIP client that my Mom can configure. Forget the fact that Skye's audio quality easily surpases any SIP client I've ever used. You bet we have to work harder to outshine Skype. I'm all over that. But we've got bigger shoes to fill than some people realize regards, Paul - Original Message - From: Mike [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 25, 2005 5:37 PM Subject: Re: [Asterisk-Users] Asterisk compare with Skype You don't get it We have to work harder to outshine Skype . : ) Its not like ford and GM. They do too very different things...we don't have to outshine them we already do. On Sat, 26 Mar 2005, Stephen wrote: Hi All, Thanks for all the comments and opinions. I think in terms of features and flexibility , Asterisk is better than Skype. But in terms simplicity, Skype is better. The problem I face is to switch Skype users to use Asterisk. Some of them use Skype for business use (on-net call) and they said Skype is enough for their business use already and find no need to use IP PBX. I think probably I need to educate them what Skype is and what Asterisk is. Maybe Asterisk community need to come up with a management system that can manage Asterisk in much simple way. We have to work harder to outshine Skype . : ) Cheers, Stephen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage Linksys Router - Life after Vonage
(On top of which, they charged me a $40 termination fee to terminate my account - just a parting shot I guess). People need to read the fine print more. From Vonage's website: If you cancel after the first 14 days of service, you will be subject to the $39.99 termination fee. If you return the device, we will refund the termination fee. The $40 termination fee isn't a stab in the back for leaving them. It's an assurance that you return their hardware. If you don't return it, you just bought it regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] IPSwitchBoard BETA
Make this another vote for Zap and IAX2 monitoring :) Paul - Original Message - From: Thorben Jensen [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 17, 2005 12:10 AM Subject: SV: SV: [Asterisk-Users] IPSwitchBoard BETA Hi Kong, No, I have no support for monitoring of Zap devices at the moment. If there is great demand for it, I will make it. Thank you Thorben -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] På vegne af Kong Sendt: 17. marts 2005 02:33 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] IPSwitchBoard BETA Is there anyway that i can add in a Zap device to be monitored? So far what i can see is, its only monitoring SIP extensions. At 05:23 AM 3/17/2005, you wrote: Very cool ... Have you tried to compile it against Mono? -JB Hawaii Thorben Jensen wrote: Fra: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] På vegne af Henry Devito Sendt: 16. marts 2005 16:17 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] IPSwitchBoard BETA I installed this and it seems to be working great. Good job. Just one question though, What is the shared extensions file? Hi Henry, The Shared extension file is a file with extension (speed dial number) that a number of users want to share, when IPSwitchBoard starts up, it will merge the extensions in the shared extension file with your extensions. You can make an extensions file by exporting extensions from IPSwitchBoard. Put this file on a shared network drive and point to that file and every time IPSwitchBoard starts up, it will merge the information in that file. regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- [This E-mail scanned for viruses by iRepublics.com Anti Virus Solutions] --- [This E-mail scanned for viruses by iRepublics.com Anti Virus Solutions] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any 24 (or 30) way FXS PCI cards?
I think first we would need to clarify the use of the term FXO in this context. I'm not a telco expert by any stretch, but it appears to me the term is used misleadingly sometimes. It seems to me that an FXO port is a port that you can plug an external phone line into - it can then allow you to dial calls out on that line, or receive calls in on that line. I see all these FXS/FXO SIP devices on the market where the FXO port is just a 'pass-through' for power-failure protection, or it allows the connected phone to dial a string that lets them bypass the PBX and use the POTS. From my minimal knowledge that isn't really FXO. Is it? I'd like to see an ethernet FXO device that does what a Zap FXO device does. If it's out there, can someone point me at it? regards, Paul - Original Message - From: Nabeel Jafferali [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 19, 2005 10:04 AM Subject: RE: [Asterisk-Users] Any 24 (or 30) way FXS PCI cards? It seems to me silly to have a T1/E1 card to connect to a channel bank when you could just have a 24/30 way FXS card in the slot in the first place. Wouldn't a SIP channel bank be better - something that has multiple FXS and FXO ports but hooks up to Ethernet. I know Wasam (ala Farfon) is try to build something like this using IAX, but is anything currently available? Nabeel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] vmware and asterisk
Anyone tried running Asterisk in production on a vmware box? I'm considering the possibility and looking for sucesses/failures... regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage a provider?
Don't sweat it - it just so happens that you came into the fray just moments after this list has had a big long drawn out argument about newbie etiquete (sp?). You've just managed to get caught in the middle. Don't let it be indicative of how everyone feels, and don't let it scare you away... :) regards, Paul - Original Message - From: Frank Abernathy [EMAIL PROTECTED] To: 'C F' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, March 11, 2005 3:49 PM Subject: RE: [Asterisk-Users] Vonage a provider? Sorry to have caused such a ruckus. It was not my intent to 'anger' someone with a noob question. I did a look-up on Google, hence me getting the information about this mail list. I am sorry that I am not a Google guru like you, so that my look-up did not get me the information I needed, so that I had to 'bother' an actual person and not some search engine. I guess some people were never noobs. :) Regardless, thank you for your response and information. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Friday, March 11, 2005 3:16 PM To: Wiley Siler; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Vonage a provider? On Fri, 11 Mar 2005 13:56:20 -0700, Wiley Siler [EMAIL PROTECTED] wrote: I don't feel I am mistreating you in asking you not to dump on a noob. Even if you do not think he is a noob and he is just lazy. You wrote: Why answer? because I don't want this to happen again. But I dont' care to help him/her or even you. Sorry this should have been: Why answer? because I don't want this to happen again. But I don't mind to help him/her or even you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.1 - Release Date: 3/9/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.1 - Release Date: 3/9/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoIP Problems?
- Original Message - From: Jay Milk [EMAIL PROTECTED] You won't be able to send caller-id NAME with any PSTN termination. That's just not how that works. Each CLEC looks up the name in some mystical database based on the phone number. How to get that DB, I don't know, but it sure would be nice to integrate something like this into *, wouldn't it? Sure, but wouldn't LiveVoip be using PRI as opposed to PSTN? I dunno about in the US, but here (Canada) we've got switch-based CallerID and user-based CallerID. As long as you're using a PRI based line the user can fire both caller number and caller name to the telco and see the results on the other side (user-based). If this isn't available in the US then that's fine, but otherwise it'd be nice to be able to send caller name along with the number through LiveVoip. regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream budgetone 101
Just pick up the handset and the speakerphone will turn off automatically. If the handset isn't hung up already, just hang it up and pick it up again. Hanging up the handset won't hang up a speakerphone call... Paul - Original Message - From: dean collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, March 07, 2005 3:21 PM Subject: [Asterisk-Users] grandstream budgetone 101 Maybe Im loosing my mind but Ive just noticed that if I put a call on speakerphone and I press speakerphone again it hangs up the call, you would expect it to take the call off speaker back on to the hand piece. Im using V 1.0.5.22 firmware. Is there any other way to turn off speakerphone Im missing? Cheers, Dean ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoIP Problems?
*shrug*. Mine's been working flawlessly since I've had it (~month). The only 2 issues I have are the ringback problem, and I can only send callerid number info to them, not name info Guess we'll see how long it lasts regards, Paul - Original Message - From: Tim [EMAIL PROTECTED] To: The Phone Guys [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, March 06, 2005 4:45 PM Subject: Re: [Asterisk-Users] LiveVoIP Problems? No. When DID go down for a whole day. Do you think thats okay? Ring busy half time or do nothing at all. Come on! Your DID's are up maybe 50% of the time if that! Why are calls failing again today? On Sun, 2005-03-06 at 17:36, The Phone Guys wrote: So you want it 100% perfect and you want it for peanuts. Makes you wonder how many *really* reliable VoIP providers there are out there? Who would you trust to handle all your incoming/outgoing business calls? Mike On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote: Anyone having problems with LiveVoIP lately? I am seeing failed outgoing calls. Calls that are being routed to wrong numbers. DID's that ring busy. For the pass 2 days I am unable to pass CID. Is anyone else have these problems? Can anyone recommend a Quality VoIP provider? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
Ummm... This isn't a Digium Run list. This is a Digium sponsored list. My understanding (someone please correct me if I'm wrong) is that this list *is not* a Digium support list. This list is a forum for Asterisk discussion by users. As such, I would suggest that all topics of discussion for all hardware manufacturers are fair game. If one isn't supposed to talk about clone cards in this forum, then this list should not be called Asterisk Users, it should be called Digium Discussion. Asterisk is an OSS product, and as such should be heavily promoted for use with all products, not just Digium's. If Digium is going to take offence to that, then they shouldn't be OSSing the software. There's plenty of discussion around non Digium hardware on this list. If you're going to get pissed off about people discussing non-Digium hardware then get pissed off about *all* of it, not just the X100P clone cards. Heck, Digium doesn't even make the X100P any more. And it's generally accepted on this list that the TDM cards have problems. So I can't say I blame people for looking for cheap alternatives. In any event, if this is a Digium Only list, then it needs to be identified as such, rather than promoted as an Asterisk list. And if that's the case, then I beg someone to start a non-Digium affilitated list so that we can have free and open discussion without worrying about getting slammed on regards, Paul - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, March 05, 2005 9:01 AM Subject: Re: [Asterisk-Users] X100P Clone, Which one? On March 5, 2005 08:14 am, Androtech wrote: I bought one Trust 56k V92 PCI Internal Modem MD-1100 which has the 1057 Motorola Chip, and I installed it on my linux box. When I try to load the module wcfxo, I cannot load it (zaptel is already loaded): Not to rub salt in the wound, but do you honestly expect the people on a Digium-run mailing list to rush out and help you after you consciously went and bought a clone card? You specifically denied Digium any income on the purchase of this hardware, and now you're asking them for help! You've got a lot of nerve. Caveat Emptor. As far as I'm concerned, you're on your own. If you're not experienced enough to figure this out on your own, you should have purchased the Dev Kit Lite, which comes with support from Digium for specifically these types of problems. Maybe someone else on this list is more forgiving than I am but I really hope not. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
- Original Message - There is a LOT of traffic on this list about products that are not supplied by Digium. Do you want to exclude those also? The Sangoma guys typically handle support for their own product, even on this list. Atacomm's card hasn't hit the market yet. The Sipura people sell hardware that Digium doesn't have similar hardware for. Very true, but I guess the point I'm trying to make is that whether or not Digium supports or runs this list, my understanding is that this list isn't intended to be a Digium hardware support forum, it's intended to be a general Asterisk Users Discussion forum. I would never expect someone to call up Digium's support channels and expect to get support for setting up a clone card. But this isn't Digium Support. The guy wasn't asking Digium to help solve his problem, he was asking 'people who use Asterisk' to help solve his problem. I'd be willing to bet there are a lot of people reading this list who have at least one clone card sitting in a server at home. It's a Users Discussion. ie. People who use Asterisk converge here to talk about Asterisk in all it's forms. Whether or not Digium runs/supports the list is beside the point. I don't really disagree with you about the cost factor - often times the pain saved by buying a $125 device is worth the extra money. However, not everyone has that luxury. While some of us (myself included) prefer to simply fix the problem by throwing money at it, a lot of people use Linux and OSS products not only because it interests them but because they're scraping out a living that doesn't let them work on anything more than the hand-me-down hardware that they've coersed off of the various people they've helped over the years, and they spend the time making it all work because time is what they've got - not money. regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
- Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] I think where the problem comes in is that people take this forum to be asterisk-biz half the time. I need X done **RIGHT NOW**!! I DEMAND HELP!! -- take it to -biz, there are dozens if not hundreds of consultants who will happily exchange money for experience. This forum is for people who want to Geepers man. Looking at the last couple of times you tried to 'educate' someone, I don't see anything in their messages that sound like I need X done **RIGHT NOW**!! I DEMAND HELP!!. Uneducated and demanding are not necessarily the same thing. Whatever, we're just going around in endless circles here. I'll get out of this discussion now and leave it up to those who wish to continue it regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [OT] - [Asterisk-Users] Why should I answer a Newbie question, therethick!
- Original Message - From: David Brodbeck [EMAIL PROTECTED] Sure. So say, I tried a Googling for X, but I didn't have any luck. Then I looked at pages X and Y in the Wiki, but couldn't find anything that related to my problem. People are a lot more sympathetic if you demonstrate you've made some effort to find the answer on your own. True, but sometimes a newbie doesn't know that people are looking for this, they're new to how lists work as well. So why not answer the question, nicely, and then say 'BTW, some people will be more symathetic if you research... yada yada...' The key thing being the term 'nicely'. Some people don't realize just how agressive their blunt approach can come across to a newbie... Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [OT] - [Asterisk-Users] Why should I answer a Newbie question,therethick!
- Original Message - From: David Brodbeck [EMAIL PROTECTED] Well, sometimes that works. But I've been on a lot of lists where newbies who thought they were being ignored started flaming people for not responding to them, writing posts badmouthing the project, hijacking other threads, accusing people of being cliquish, etc. Sometimes you just can't win. Sure. But if they flame, then they deserve to get hammered on. In that case, hammer away. However, I don't think it's fair to lump all newbies into that basket. Let them throw the first punch, rather than assume that all newbies who don't know better are out to wreak havoc How would the experts like it if everyone assumes they're a bunch of arrogant techies who only want to talk down to those less worthy, before they speak up and prove it to be true? :) Don't make the same assumptions in the reverse for the newbies... regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
Ok, time for me to ask my own newbie question. :) I've done some digging on ringback, and if I'm understanding it correctly, it's the ring tone that the caller hears when dialing another person. What exactly is it that people are finding now working with LiveVoip? Everyone says 'ringback isn't working', but nobody's really explained exactly what's happening. At least not that I've been able to find. I have a DID with them, and it works just fine. Dialing out works fine, when people call in it works fine. I'm interested in knowing what it is that isn't working, and if I can re-create it on my system... regards, Paul - Original Message - From: Ed Greenberg [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2005 11:12 AM Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip --On Friday, March 04, 2005 11:58 AM -0600 James Taylor [EMAIL PROTECTED] wrote: It would be nice if they told us what the problem with Asterisk is... There's probably enought great minds on this list, that it could be resolved. There is clearly an issue between LiveVoip and Asterisk. The LiveVoip people claim that they have been ignored on the Asterisk List and they indeed blame Asterisk for everything from lost dtmf to other failures. That said, they are the only company I've found that offers inbound DIDs with multiple simultaneous calls, suitable for a call center or calling card application. Most others limit you to one, or a small few, inbound paths. They (Level 3, actually) also have the widest coverage for DIDs in the US. At the current level of service, LiveVoip is not going to get my business. If I can find anybody else to provide my inbound service, I'm very interested in talking to them. /edg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
- Original Message - From: Ryan Laginski [EMAIL PROTECTED] Anyways, I haven't found anyone that offers a toll free number that works in Canada for 1.29 cents a minute. If there is others, please let me know. You're LiveVoip toll free number costs 1.29 c/min from Canada? My toll free number through LiveVoip costs me 5 c/min when calling from Canada (1.2 c/min from US). Hmm wonder what I need to do to get that deal... :) Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
Hi Robert, I tried yours and Steven's scenarios and you're absolutely right. I get ringback when the initial call takes place, but if I then try to do a transfer to another extension after the fact I do not hear ringback on the line. So I absolutely agree that there is a problem. What I'm not trying to understand is how Ringback works in this context. For lack of knowing better, my first thought would be that LiveVoip would be correct - that the problem is with asterisk, since I would have assumed that once LiveVoip has connected the call and asterisk has answered, all they're doing is providing audio in and out - wouldn't be Asterisk's responsibility to provide new ringtones to the calling party at this new transfer point? However, if everyone *does* get ringback when using other providers then it makes sense that there's something happening at LiveVoip's end. *shrug*. I'm interested in what the technicals are here regards, Paul - Original Message - From: Robert Webb [EMAIL PROTECTED] To: Paul Fielding [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2005 12:06 PM Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip On Fri, 04 Mar 2005 11:46:27 -0700 Paul Fielding [EMAIL PROTECTED] wrote: Hmmm. My server is currently set to let the line ring for 20 seconds, ringing several extensions internally. (I do not answer the line, it just rings the extensions). If I don't pick up after 20 seconds it then answers the line and sends to voicemail or to an auto-attendant, depending on the situation. Ringback seems to be working for me, I hear ringing on the calling end... *shrug*. Paul Ok,I have to retract my last statement and give an update. It has been a while since I had played with the DID I have from them. It is not an issue before the * box picks up. I set my incoming context to ring my VoIP phone for 20 seconds directly with using the IVR system and I had the ringing. But when I restored it to no background on hold music and issued a dial command of Dial(SIP/2001,15,r) instead of Dial(SIP/2001,15,m), after the IVR plays its intro, I got no ringing on the calling end. Just dead air from LiveVoIP. I then used this same test context by dialing in through a VP Connect account and after the initial greeting and moving to the Dial command, I got the ringing on the the calling end. Sorry for the incorrect info the first time, it had just been quite a while since I had played with the Live account. Robert - Original Message - From: Robert Webb [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Friday, March 04, 2005 11:42 AM Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip On Fri, 04 Mar 2005 11:35:55 -0700 Paul Fielding [EMAIL PROTECTED] wrote: Ok, time for me to ask my own newbie question. :) I've done some digging on ringback, and if I'm understanding it correctly, it's the ring tone that the caller hears when dialing another person. What exactly is it that people are finding now working with LiveVoip? Everyone says 'ringback isn't working', but nobody's really explained exactly what's happening. At least not that I've been able to find. I have a DID with them, and it works just fine. Dialing out works fine, when people call in it works fine. I'm interested in knowing what it is that isn't working, and if I can re-create it on my system... regards, Paul Setup your * box to not answer the call right away. Allow for say 5 seconds of ringing. Then call into it on one of your DID's. From the calling end all you will get is dead air. No ringing. At least this is the issue I am having.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
What I'm not trying to understand is how Ringback works in this context. err, I mean what I'm now trying to understand. Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [OT] - [Asterisk-Users] Why should I answer a Newbie question,therethick!
- Original Message - Look, don't answer lame questions if you don't want to. Flaming a newb for being a newb is just mean. (they will eventually RTFM or STFW or they will fail). This is the way of the open source community. Here Here, I'm with you. I find it a constant source of amazement how, in all the various lists I've followed, people find it necessary to beat on the new guy. Even the 'if you don't want to get flamed then do some research first' attitude i'm not a fan of. Sometimes newbies are also newbies to the concept of lists, etc, as well as the topic of the list. Frankly, I agree. If you don't like the question, feel it's lame or dumb, or don't like that someone hasn't done their research, then delete the message. If you think they're wasting your time by writing a message, then don't waste any more of your own time by responding to it. I find the pummelling of newbies more annoying than the newbie question itself. regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does the g.729 registration program work?
I thought that it was 3 different times with different nics, but with the same nic didn't count.*shrug*. No matter - if we can just copy the license keys that's much easier... :) Paul - Original Message - From: Kristian Kielhofner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 28, 2005 2:39 AM Subject: Re: [Asterisk-Users] How does the g.729 registration program work? Paul Fielding wrote: I could be mistaken, but doesn't the license tie itself to the nics on the server? I believe the Digium server will allow you to reregister as much as you want as long as it's still got the same nics... Paul You do not need to re-register a key if the NICS have not changed. Just make a copy of the files in /var/lib/asterisk/licenses (I think that's where they are). Digium will allow you to re-register the same G729 license key three different times without having them reset your license key. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does the g.729 registration program work?
- Original Message - From: Kristian Kielhofner [EMAIL PROTECTED] Paul Fielding wrote: I thought that it was 3 different times with different nics, but with the same nic didn't count.*shrug*. No matter - if we can just copy the license keys that's much easier... :) Yes, that is the case. But on the same hardware why not just copy the license files? Don't get me wrong, I agree with you. I'm just pointing out that if you aren't in a position to copy the license files (ie. you blew up the drive or simply forgot) then you're still ok regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does the g.729 registration program work?
You misunderstand. Ofcourse I need to run the register program on the machine itself. The point is I build them from images and every now and then I roll out a new image. My question is, what do I need to preserve from the previous image to keep the licences. Obviously reformatting the disk and reregistering is not going to work. I could be mistaken, but doesn't the license tie itself to the nics on the server? I believe the Digium server will allow you to reregister as much as you want as long as it's still got the same nics... Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom phone hint exten question
- Original Message - From: James Bean [EMAIL PROTECTED] I am going to now sit in a corner and go quietly insane while playing the banyo with no strings. I'd probably go insane, too, if I was trying to figure out how the heck to play a banyo ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM-400P Sound Quality issues
I'm running a TDM-400P with 2 x FXS and 2 x FXO. I'm finding that there seems to be an odd relationship to sound quality on the card to my local when connecting via a SIP client. When I'm on my local network, if I connect to Asterisk via a SIP client (such as x-pro), and dial an outside line through the card, sound quality seems quite good. However, when I'm at a remote location and connect via the same SIP client and dial an outside line, the audio quality is fuzzy, sometimes quiet, and generally more difficult to understand. I spent a bunch of time troubleshooting the SIP end of things, thinking that's where the problem was, until I realized that every other SIP connection I make (from remote) yields a high quality call. ie. I can dial another SIP client and maintain high quality audio. Additionally, I can dial an extension that not only SIP connects to my server, but from there goes out an IAX2 connection to another remote Asterisk server, from there to another SIP client, and the audio quality is excellent. Therefore, I don't think the audio issue I'm experiencing is on the SIP end. Are there some wierd SIP - ZAP timing / conversion / other issues that could be causing this? thoughts? regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM-400P alternatives?
Are there any other relatively low cost analog cards available? I'm interested in finding something that might work a bit more reliably than the TDM-400P regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM-400P alternatives?
- Original Message - From: Jon Gabrielson [EMAIL PROTECTED] You didn't say what your fxs/fxo requirements are but: A T1 card ($500) and a used channel bank ($300) might be a good alternative. Basically my fxs/fxo requirements are the same as my existing TDM-400P ( 2 in 2 out). Just trying to find something that works more reliably than this card has turned out to be. Paul Cheers, Jon. On Sunday 13 February 2005 02:55 pm, Paul Fielding wrote: Are there any other relatively low cost analog cards available? I'm interested in finding something that might work a bit more reliably than the TDM-400P regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM-400P Sound Quality issues
Pure guess... you're probably bumping into some of the same issues that many of us TDM users are hitting. Seems like either an interrupt handling (latency) or pci bus issue. You'll find hundreds of postings relative to this over the last six months or so. Not everyone has problems with the TDM, but some have found that swapping motherboards does clear up the issue. I did a bunch of searching through the list, found lots of messages regarding misc. TDM400p problems, but none that sounded like the issue I'm seeing. Can anyone point me to any discussions regarding this? The thing that I find so odd about it is that the sound quality only degrades on the zap channel when I'm connecting from a *remote* SIP client, but on local network the zap channel sounds fine (see description below). I'm willing to get a different MB if that's really the fix, but I'd hate to go through the work and $$ to make that happen only to find that the problem doesn't go away... Paul - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, February 13, 2005 9:12 PM Subject: Re: [Asterisk-Users] TDM-400P Sound Quality issues I spent a bunch of time troubleshooting the SIP end of things, thinking that's where the problem was, until I realized that every other SIP connection I make (from remote) yields a high quality call. ie. I can dial another SIP client and maintain high quality audio. Additionally, I can dial an extension that not only SIP connects to my server, but from there goes out an IAX2 connection to another remote Asterisk server, from there to another SIP client, and the audio quality is excellent. Therefore, I don't think the audio issue I'm experiencing is on the SIP end. Are there some wierd SIP - ZAP timing / conversion / other issues that could be causing this? thoughts? Pure guess... you're probably bumping into some of the same issues that many of us TDM users are hitting. Seems like either an interrupt handling (latency) or pci bus issue. You'll find hundreds of postings relative to this over the last six months or so. Not everyone has problems with the TDM, but some have found that swapping motherboards does clear up the issue. Processor speed and ram have nothing to do with it, nor does single vs dual processors, etc. Several people have opened trouble tickets with digium, but seems all have gone into a black hole (thus far). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P FXO lines problem
I've seen the same behavior on my TDM400P. I solved it by simply scripting a stop/start of the zaptel drivers and asterisk in the middle of the night each night. Of course, that might not be practical in a more seriously production environment Paul - Original Message - From: Micha Mosiewicz [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 09, 2005 6:52 PM Subject: [Asterisk-Users] TDM400P FXO lines problem We are experiencing problems with FXO modules on TDM400P. From time to time they stop responding to incoming rings although they work fine if we use them to dial out. It's been verified at least in two different installations (using different mainboards) in two different locations. The only solution to the problem is to stop asterisk, then unload and reload kernel drivers. The problem appears since we started to use asterisk (September) till now, although we have tried diffrent versions. 26 days ago we updated them to CVS version and today the problem reappeared. Average time between failures seems to be around 20-30 days. At first we thought it might be caused becouse we had 5 TDM400P cards in one server. But the problems was also spotted in machines that had only one TDM400P (2FXO/2FXS) card. -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy's apparantly failing in the field
We have deployed many (20+) IAXy's in the field. At a couple of locations, the IAXy's have just stopped working after 1 or 2 days use. No lights go on, no DHCP lease is renewed as far as we can tell, and of course no dialtone and no registration with the server. I bought two of them, both of them experienced problems with 2 weeks, one died completely within 3. I'm going back to using Grandstream 286 devices. Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] top-notch servers/OS/network, ulaw codec - sound still choppy
- Original Message - I see the sip user is an external ip. I would take a look at your QoS settings on your router. Make sure the voice traffic is getting the priority it deserves. Also, check for packet loss. I'd still be wondering if there's something else. I, too, experience choppy SIP connectivity from external IPs, but as I've mentioned in previous postings, I have a Vonage ATA that seems to have no problems keeping a crystal clear connection as it leaves my place and goes to Vonage's servers, so I think there must be more to it than QoS. I have to believe that there's some more jitter correction or other such buffering that could berhaps be played with, though I don't know what it would be *shrug*.? regards, Paul On Thu, 20 Jan 2005 11:26:02 -0800, Gabriel Afana [EMAIL PROTECTED] wrote: Hi, My SIP calls are sounding a little choppy. I've did my research but everything looks right on my end...what am I missing? Running RedHat ES 3.0 on dual AMD Opteron servers. My system is cololocated in downtown LA and is fed via a gigabit handoff from XO, ATT, Level 3 and Wiltel (I have a 100Mb didicated line). So I dont think its the Servers, its the network, Asterisk is working fine and all codecs look right...what could be the cause? **SNIP FROM SIP.CONF*** [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) ;bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records allow=ulaw ; Allow codecs in order of preference * ga0*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format 64.201.99.2479092479878 2fd496bf330 00103/00105 ulaw ga0*CLI show version Asterisk 1.0.3 built by [EMAIL PROTECTED] on a x86_64 running Linux P.S. in my sip.conf file, it looks like I am only allowing the ulaw codec...could that cause a problem if I happen to need to call somebody that doesn't support ulaw? Gabe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault after Digitnetwork X100Pinstall
Shouldn't you contact your vendor for support and not a different vendors support channel? Um, I didn't think this was a Digium support channel. I thought this was an Asterisk Users channel. Seems to me the question should be fair game. (Sorry I don't have an answer to your question, though, Dave). regards, Paul - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 20, 2005 9:41 PM Subject: Re: [Asterisk-Users] Segmentation Fault after Digitnetwork X100Pinstall On Fri, 2005-01-21 at 16:57 +1300, Dave Green wrote: I've just installed a Digitnetworks X100P clone in my * server and run the install script for the voicepet-single-x100p tarball. The install appeared to run OK with modprobe wcfxo successful and the ztcfg reporting Channel 01: FXS Kewlstart (Default) (Slaves: 01). When I try to start * though I get a segmentation fault after loading res_features.so. I discovered that the Digitnetwork install script seems to modify all of the .conf files, leaving .conf.old copies. I tried moving the .conf.old files back to .conf but am still get the seg fault. Shouldn't you contact your vendor for support and not a different vendors support channel? Another company with retarded disclaimers sending to a KNOWN publicly archived mailing list. Fix the problem or be ridiculed regularly for the stupidity. CAUTION: This message and any attachments contain privileged and confidential information. If you are not the intended recipient of this message, you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately via email and then destroy this message and any attachments. Any views expressed in this message are those of the individual sender and may not necessarily reflect the views of Winstone Pulp International Ltd. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729? Worth it?
Low bandwidth Low CPU utilization Best audio quality I think you might want to clarify that Best audio quality is in relation to other highly compressed codecs. Certainly my (albeit limited) experience is that g711 is much more clear than g729. Compared against gsm, for example, however, the audio quality is quite good regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best Grandstream firmware to use?
I've seen lots of stuff go around about Grandstream firmware levels (in my case specifically the BT101/102). I'm just wondering what the currently accepted 'best' firmware version is to use? After seeing stuff going around about buggy firmware I want to know what I'm getting into before upping past my current 1.0.5.11. It's relatively stable, and the last thing I want to do is update to a flaky firmware Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound quality - commercial vs. Asterisk
- Original Message - From: Sean Kennedy [EMAIL PROTECTED] Likely, you are running into packet queue problems. As I recall, the vonage device goes on the line before anything else, so it can shape the stream to put it's bits first, ensuring it's packets get out in a timely matter ( #1 important thing in voip ). If you were to shape your stream and put your voip bits first, then I think you'd see an improvement in the qualty of service. I agree I probably am having some packet queue problems, however i don't think it's my only problem. My Vonage ATA adapter is actually further behind the line than my Asterisk server. My configuration is such: Cable Modem -- Asterisk Server Linux Router (Each have their own real IP) -- Vonage ATA (behind Linux router) I don't have QoS running on my Linux box, though I've been thinking about trying to implement it. If I did manage to get it implemented then I'll probably also move my Asterisk server behind the router. Up until now I've left the Asterisk server in the real world due to problems running Asterisk behind a NAT. Those problems, however, seem to have been dealt with and a friend of mine is successfully running his behind a NAT.So QoS may be the way to, perhaps. I don't think it'll resolve all my issues though - if the Vonage ATA can do it withough QoS running, then surely there's more I can do with Asterisk. Perhaps the new jitter buffering coming soon will fix... *shrug*... Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do i talk to the IAXy...? (Newbie Alert)
I've actually given up on my two IAXy devices - one of them keeps loosing it's connection and needs to be rebooted, the otherone keeps its connection but periodically loses it's ability to send clear audio and needs to be rebooted. I'm going back to the Grandstream ATAs that work just dandily for me, and are cheaper, to boot. IMHO the IAXy is not yet ready for production use. They're going to go into a box and sit there until a firmware update fixes the problems or until someone braver than me offers to buy them off of me for a cut rate :) regards, Paul - Original Message - From: blackburn [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 18, 2005 7:38 AM Subject: Re: [Asterisk-Users] How do i talk to the IAXy...? (Newbie Alert) Hi. Due to the lack of any instructions from Digium, I have bought this IAXy but have been completely unable to use it at all. Now that I've tried provisioning my IAXy using the software given below. I get the message Got response back from 192.168.0.3. Does this mean that it has been successfully provisioned? But when I pick up my phone, there is still no dial tone. What am I still missing??? How do I know if my IAXy has been properly provisioned? Thanks!! Daiku wrote on 2005/01/18 20:59: Quoting from message: 05/01/18 20:24 +0900 sent by Leonardo Gomes Figueira: If you don't wanna install cygwin you can download iaxyprov compiled with Cygwin from: ftp://ftp.planetarium.com.br/pub/util/voip/iaxyprov/ Hi, and thanks for the hint...! A few days ago another helpful list subscriber sent me a small utility that he compiled and tested himself, but since that day i have not been at the place yet where i can use the Windows machine. Sometime this week or next week... I'll report the results here when all is done. Thanks regards: H. D. -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound quality - commercial vs. Asterisk
- Original Message - From: Brian Capouch [EMAIL PROTECTED] Paul Fielding wrote: I agree I probably am having some packet queue problems, however i don't think it's my only problem. My Vonage ATA adapter is actually further behind the line than my Asterisk server. My configuration is such: I wonder about codec-related issues, as well. Any chance that the Cisco adapter is using g711 and the other phone is using something with a lesser sound quality, or that perhaps some transcoding is going on that might be introducing some quality degradation? I'm not sure what codec the Vonage ATA (Linksys) is using, though I'd be interested to find out. When using my Asterisk server, either using Grandstream BT-101/2 or X-Ten Pro, I'm using g711. In some cases it's g711 to g711, in other's it's g711 to a ZAP chanel to an outside line. I don't think there's any other transcoding going on Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transfers with zap channel
Ok, I've seen discussion before on doing transfers
(attended and unattended), there seems to be much confusion over
it.
As things sit, I've been trying (unsuccessfully) to
do transfers with a zap channel (analog phone attached to TDM400). I have
no idea what I'm missing. My current understanding is that I need to have
transfer=yes in zapata.conf, do a flash hook, dial the 2nd number, flash hook
again and we're linked (attended). Then if I hang up the call will
be transfered.
However, when I try to do this things don't
work. Here's what I do:
- connection is made between Zap/3 (analog phone)
and Zap/1 (outside line).
- flash hook to get dialtone (I do get
dialtone)
- attempt to transfer to extension 7007 - I dial
7007
- after dialing the 2nd zero, and before dialing
the 2nd seven, I hear Asterisk announce (seven - zero - one) and then Zap/3 is
hung up (I get a busy signal). Zap/1 gets parked.
Here's what the log shows:
-- Zap/1-1 answered
Zap/3-1 -- Attempting native bridge of Zap/3-1 and
Zap/1-1 -- Started three way call on channel
1 -- Started music on hold, class 'default', on
Zap/3-1 -- Attempting native bridge of Zap/3-1 and
Zap/1-1 -- Starting simple switch on
'Zap/1-2' -- Started music on hold, class 'default', on
Zap/3-1 == Parked Zap/3-1 on 701. Will timeout back to dostuff,7001,1
in 45 seconds -- Added extension '701' priority 1 to
parkedcalls -- Playing 'digits/7' (language
'en') -- Hungup 'Zap/1-1' == Spawn extension
(dostuff, 7001, 1) exited non-zero on
'Parked/Zap/3-1ZOMBIE' -- Stopped music on hold on
Parked/Zap/3-1ZOMBIE -- Playing 'digits/0'
(language 'en') -- Playing 'digits/1' (language
'en') -- Parking call to 'Zap/1-2'
-- Hungup 'Zap/1-2' -- Stopped music on hold on
Zap/3-1 == Zap/3-1 got tired of being parked --
Hungup 'Zap/3-1'
I'm not sure what I'm missing. Apparently
something to do with parked calls, so I must be misunderstanding how do to the
call transfer.
I've also tried enabling Asterisk transfer on the
channel (exten = 7010,1,Dial(${CORDLESS},20,tT)).
My understanding of this method is that this allows
one to hit the pound (#) to start a transfer. Yet pound does
nothing. Is it fair to assume that the tT only works on SIP channels, or
am I missing something else.
Any help is much appreciated
Paul
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Re: [Asterisk-Users] internal dial tone on password from outside
When I experimented with DISA, I found it to be very unreliable - sometimes it would ignore my key presses and just keep giving dialtone, sometimes it would work. I couldn't find a rhyme or reason to it. I ended up just giving up and going with the silence Paul - Original Message - From: Brian Dingman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 17, 2005 7:43 PM Subject: Re: [Asterisk-Users] internal dial tone on password from outside http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] transfers with zap channel
The outside line isn't actually being dropped - the
outside line hanging up is me hanging up the outside line after finding that my
transfer failed.
I must be not understanding how the flash-hook
works then. My understanding was that when I flash-hook and get a second
dialtone I should be able to dial the extention I want to reach (7007 is another
extension, via SIP). Normally, if I pick up the analog phone and dial 7007
it rings the extention fine. Apparently, though, when you get that
second dialtone, it has different rules? I haven't been able to find
documentation on this, can it be found anywhere? For example, why does
dialing 700 park the call? I haven't found anything on this...
*shrug*...
Paul
- Original Message -
From:
Lyle
Giese
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, January 17, 2005 7:22
PM
Subject: Re: [Asterisk-Users] transfers
with zap channel
How long between getting parked is the orginal
call dropping?
Depending on your dialplan, yes dialing 700x will
almost immediately send the call to call parking. (IMHO, poor extension
planning can also cause this.)
I don't use the t or T
optionsPERIOD. IMHO, you just lose the ability to use the # key
and confused the heck out of my users. Took it out and use the flash
method only in my dial plan. Dial 700, park the call. Dial the
other extension, tell them to pick up 701. Or use meetme for conference
calling?
I know I need to play with three way calling here
also.
Lyle
- Original Message -
From:
Paul
Fielding
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, January 17, 2005 6:12
PM
Subject: [Asterisk-Users] transfers
with zap channel
Ok, I've seen discussion before on doing
transfers (attended and unattended), there seems to be much confusion over
it.
As things sit, I've been trying
(unsuccessfully) to do transfers with a zap channel (analog phone attached
to TDM400). I have no idea what I'm missing. My current
understanding is that I need to have transfer=yes in zapata.conf, do a flash
hook, dial the 2nd number, flash hook again and we're linked
(attended). Then if I hang up the call will be
transfered.
However, when I try to do this things don't
work. Here's what I do:
- connection is made between Zap/3 (analog
phone) and Zap/1 (outside line).
- flash hook to get dialtone (I do get
dialtone)
- attempt to transfer to extension 7007 - I
dial 7007
- after dialing the 2nd zero, and before
dialing the 2nd seven, I hear Asterisk announce (seven - zero - one) and
then Zap/3 is hung up (I get a busy signal). Zap/1 gets
parked.
Here's what the log shows:
-- Zap/1-1 answered
Zap/3-1 -- Attempting native bridge of Zap/3-1 and
Zap/1-1 -- Started three way call on channel
1 -- Started music on hold, class 'default', on
Zap/3-1 -- Attempting native bridge of Zap/3-1 and
Zap/1-1 -- Starting simple switch on
'Zap/1-2' -- Started music on hold, class 'default',
on Zap/3-1 == Parked Zap/3-1 on 701. Will timeout back to
dostuff,7001,1 in 45 seconds -- Added extension '701'
priority 1 to parkedcalls -- Playing 'digits/7'
(language 'en') -- Hungup 'Zap/1-1' == Spawn
extension (dostuff, 7001, 1) exited non-zero on
'Parked/Zap/3-1ZOMBIE' -- Stopped music on
hold on Parked/Zap/3-1ZOMBIE -- Playing
'digits/0' (language 'en') -- Playing 'digits/1'
(language 'en') -- Parking call to
'Zap/1-2' -- Hungup 'Zap/1-2' --
Stopped music on hold on Zap/3-1 == Zap/3-1 got tired of being
parked -- Hungup 'Zap/3-1'
I'm not sure what I'm missing. Apparently
something to do with parked calls, so I must be misunderstanding how do to
the call transfer.
I've also tried enabling Asterisk transfer on
the channel (exten = 7010,1,Dial(${CORDLESS},20,tT)).
My understanding of this method is that this
allows one to hit the pound (#) to start a transfer. Yet pound does
nothing. Is it fair to assume that the tT only works on SIP channels,
or am I missing something else.
Any help is much appreciated
Paul
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Re: [Asterisk-Users] transfers with zap channel
Ah, suddenly everything becomes clear. I've never looked in features.conf before. I now understand that 700 is supposed to intitiate the call park, and it's taking precidence over the extension I was trying to dial of 7007. I've changed the call parking extension and now I can do regular attended and unattended transfers without having to park the call... (note to anyone else changing features.conf, you have to 'restart' asterisk, a 'reload' won't do). thanks a bunch for the help, guys... Paul - Original Message - From: Lyle Giese To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 17, 2005 8:20 PM Subject: Re: [Asterisk-Users] transfers with zap channel Have you looked at features.conf? Lyle - Original Message - From: Paul Fielding To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 17, 2005 8:53 PM Subject: Re: [Asterisk-Users] transfers with zap channel The outside line isn't actually being dropped - the outside line hanging up is me hanging up the outside line after finding that my transfer failed. I must be not understanding how the flash-hook works then. My understanding was that when I flash-hook and get a second dialtone I should be able to dial the extention I want to reach (7007 is another extension, via SIP). Normally, if I pick up the analog phone and dial 7007 it rings the extention fine. Apparently, though, when you get that second dialtone, it has different rules? I haven't been able to find documentation on this, can it be found anywhere? For example, why does dialing 700 park the call? I haven't found anything on this... *shrug*... Paul - Original Message - From: Lyle Giese To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 17, 2005 7:22 PM Subject: Re: [Asterisk-Users] transfers with zap channel How long between getting parked is the orginal call dropping? Depending on your dialplan, yes dialing 700x will almost immediately send the call to call parking. (IMHO, poor extension planning can also cause this.) I don't use the t or T optionsPERIOD. IMHO, you just lose the ability to use the # key and confused the heck out of my users. Took it out and use the flash method only in my dial plan. Dial 700, park the call. Dial the other extension, tell them to pick up 701. Or use meetme for conference calling? I know I need to play with three way calling here also. Lyle - Original Message - From: Paul Fielding To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 17, 2005 6:12 PM Subject: [Asterisk-Users] transfers with zap channel Ok, I've seen discussion before on doing transfers (attended and unattended), there seems to be much confusion over it. As things sit, I've been trying (unsuccessfully) to do transfers with a zap channel (analog phone attached to TDM400). I have no idea what I'm missing. My current understanding is that I need to have transfer=yes in zapata.conf, do a flash hook, dial the 2nd number, flash hook again and we're linked (attended). Then if I hang up the call will be transfered. However, when I try to do this things don't work. Here's what I do: - connection is made between Zap/3 (analog phone) and Zap/1 (outside line). - flash hook to get dialtone (I do get dialtone) - attempt to transfer to extension 7007 - I dial 7007 - after dialing the 2nd zero, and before dialing the 2nd seven, I hear Asterisk announce (seven - zero - one) and then Zap/3 is hung up (I get a busy signal). Zap/1 gets parked. Here's what the log shows: -- Zap/1-1 answered Zap/3-1 -- Attempting native bridge of Zap/3-1 and Zap/1-1 -- Started three way call on channel 1 -- Started music on hold, class 'default', on Zap/3-1 -- Attempting native bridge of Zap/3-1 and Zap/1-1 -- Starting simple switch on 'Zap/1-2' -- Started music on hold, class 'default', on Zap/3-1 == Parked Zap/3-1 on 701. Will timeout back to dostuff,7001,1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls -- Playing 'digits/7' (language 'en') -- Hungup 'Zap/1-1' == Spawn extension (dostuff, 7001, 1) exited non-zero on 'Parked/Zap/3
[Asterisk-Users] Sound quality - commercial vs. Asterisk
So far in my playing with Asterisk I've messed with soft phones (x-ten, sjphone), hard phones (Grandstream 102), and ATA adapters (Grandstream 286, Digium IAXy). I've also got a Vonage line, using a Linksys ATA. None of the devices I've connected to my Asterisk server have been able to maintain the same consistent sound quality over a long distance as the Vonage line. Don't get me wrong, the Grandstreams are actually not too bad, but there is still some breakups that can be annoying. Meanwhile the Vonage ATA maintains an almost flawless connection, all the time. I'm assuming (perhaps wrongly?) that the Linksys ATA that Vonage uses is still using SIP with some standardized codec. If that assumption is correct, then how the heck to they manage to get the consistent connection quality? Is it just a matter of the right setting tweaks within Asterisk and/or the SIP devices? I don't think it's a question of Asterisk hardware, since if I connect via local network to the Asterisk server with a SIP device the quality is pretty consistent. It's generally when remotely connecting that I have the inconsistent sound quality. This would lead me to believe that it's a matter of tweaking something to deal with latency or packet dropping issues (?). What has Vonage got figured out that I still need to? Any comments would be appreciated... regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream Bugetone 101 mwi
Hahawell the MWI is the blinking blue LCD. The message button is reserved for future use Hang in there. There will soon to be some upgrades and rumor has it that the conferencing feature will soon be introduced so that conference button on the phone will soon be working. The message button isn't reserved, it works fine, you simply need to correctly configure it. It's job is to dial the voicemail box when pressed. This works as designed. It just doesn't blink. Paul David On Fri, 14 Jan 2005 10:25:46 -0500, Stephen R. Besch wrote Ronald Wiplinger wrote: I tried to use message waiting indicator, by Subscribe for MWI in the web menu of the phone. However, it does not light up / flash, even if a voice mail is waiting. Where is the switch to turn it to? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I don't mean to be rude to everyone who responded to this question, but I think that everyone is answering the wrong question. The point is that the message waiting indicator doesn't light up, at all, ever. All that happens when messages are waiting is that the display blinks and the phone gives a stutter dialtone. That's it. There is no light under the button - there should be, but there isn't. The blinking phone designers should have put those stupid blinking red leds - that only flash on boot up - under the message button and flashed the display during boot up. But they didn't and we're stuck with it. Such is life. Stephen R. Besch ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - David Liu Chief Operating Officer Deltapath Commerce Technology Limited HK Tel: +852 3107-1333 HK Cell: +852 9166-1880 US Tel: +1 313 228-0906 - The Linux Enterprise Technology Provider! www.deltapath.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Bugetone 101 mwi
[EMAIL PROTECTED] It occurs to me, do you have the numbered extension set the same as the context name for the phone in sip.conf? For example, in my sip.conf, the context names for each phone are [7001], [7002] etc. However, this doesn't necessarily need to be true. If it's not true, try: mailbox=context name in sip.conf instead. so for example, if the extension is 7123, but the context name for the phone in sip.conf is fred, try: mailbox=fred Just a shot in the dark Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for a wireless phone... wifiortraditional wireless ?
You shouldn't need any port forwarding. I've found any SIP phones I've worked with have happily moved from site to site behind NATs, etc. So I see no reason to believe that the WiFi phone would be any different - it's just connecting wirelessly instead of with a wire. And to answer your second question, I suspect some people would have very good use for such a phone. Myself personally, I travel often - in the last 6 weeks I've been home for precisely 4 days. Most of the hotels, etc I stay at have wireless of one sort or another, most of my friends have wireless in their homes, may offices I work at have wireless.So I could see very good use for having a wifi phone I can take with me regards, Paul - Original Message - From: Brian Johnson [EMAIL PROTECTED] To: Asterisk Users Mailing List -Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 13, 2005 7:41 AM Subject: Re: [Asterisk-Users] Looking for a wireless phone... wifiortraditional wireless ? In that example you could make outgoing calls only correct? (since incoming likely needs port forwards) I guess the questions becomes how often are you going to do that to justify the extra $100 or so you going to pay for a wifi sip phone? Paul Fielding ([EMAIL PROTECTED]) wrote: I think some people are missing the point. You can't throw your cordless phone in your pocket, go to your office, hotel or buddie's house, turn it on and get a signal. You can with a WiFi phone, however - Original Message - From: Kim Lux [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 12, 2005 10:01 PM Subject: Re: [Asterisk-Users] Looking for a wireless phone... wifi ortraditional wireless ? I don't know why people keep making the statement about range with wifi versus a cordless phone. I can easily get a good wifi signal when I'm over at my neighbors but can't get reception with our cordless (2.4GHz) phone. (Both receivers are at home...) It seems to me that the wifi range is at least as good as the phone range if not better. Thanks for the tip on the SIP phones. On Wed, 2005-01-12 at 18:53 -1000, James H. Thompson wrote: Uniden and Vtech both just announced cordless phones with SIP ATAs built into the base station. You get better range and battery life compared to a WiFi phone. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Kim Lux To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, January 12, 2005 5:49 PM Subject: Re: [Asterisk-Users] Looking for a wireless phone... wifi or traditional wireless ? An unflattering zyxel review: http://slacker.com/~nugget/asterisk3.php I can't help but think my questions are out of place on this list... I'm asking questions about SIP phones and everyone else is talking about asterisk. Sorry. On Wed, 2005-01-12 at 20:08 -0700, Kim Lux wrote: My wife wants a cordless phone for around the house. We are going to be using VOIP exclusively very shortly. Our current cordless phone is aged and on the verge of replacement. The other phone we are going to use is a SIP Budgetone. Should I buy a SIP to POTS converter and a new cordless phone or a wifi SIP phone ? Is anyone using the Pulver WiSIP phone ? Any comments ? How about the zyxel ? Thanks -- Kim Lux, Diesel Research Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kim Lux, Diesel Research Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing
Re: [Asterisk-Users] Grandstream Bugetone 101 mwi
- Original Message - From: Eric Wieling aka ManxPower [EMAIL PROTECTED] the mailbox= has NOTHING to do with extensions.conf at all. [EMAIL PROTECTED] voicemail.conf: [happypeople] 666 = 1234,Happy Dude sip.conf [blah} ... [EMAIL PROTECTED] Boy, I had a blonde moment back there, I was shooting from the hip and in looking at my response realize the error.The one thing I am wondering about, though, is the need for specifying context. I'm not specifying any context in my mailbox= line and everything works fine. Then again, I only have one context in my voicemail.conf file... *shrug* Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for a wireless phone... wifi ortraditional wireless ?
I think some people are missing the point. You can't throw your cordless phone in your pocket, go to your office, hotel or buddie's house, turn it on and get a signal. You can with a WiFi phone, however - Original Message - From: Kim Lux [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 12, 2005 10:01 PM Subject: Re: [Asterisk-Users] Looking for a wireless phone... wifi ortraditional wireless ? I don't know why people keep making the statement about range with wifi versus a cordless phone. I can easily get a good wifi signal when I'm over at my neighbors but can't get reception with our cordless (2.4GHz) phone. (Both receivers are at home...) It seems to me that the wifi range is at least as good as the phone range if not better. Thanks for the tip on the SIP phones. On Wed, 2005-01-12 at 18:53 -1000, James H. Thompson wrote: Uniden and Vtech both just announced cordless phones with SIP ATAs built into the base station. You get better range and battery life compared to a WiFi phone. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Kim Lux To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, January 12, 2005 5:49 PM Subject: Re: [Asterisk-Users] Looking for a wireless phone... wifi or traditional wireless ? An unflattering zyxel review: http://slacker.com/~nugget/asterisk3.php I can't help but think my questions are out of place on this list... I'm asking questions about SIP phones and everyone else is talking about asterisk. Sorry. On Wed, 2005-01-12 at 20:08 -0700, Kim Lux wrote: My wife wants a cordless phone for around the house. We are going to be using VOIP exclusively very shortly. Our current cordless phone is aged and on the verge of replacement. The other phone we are going to use is a SIP Budgetone. Should I buy a SIP to POTS converter and a new cordless phone or a wifi SIP phone ? Is anyone using the Pulver WiSIP phone ? Any comments ? How about the zyxel ? Thanks -- Kim Lux, Diesel Research Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kim Lux, Diesel Research Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Bugetone 101 mwi
you need to set 'mailbox=extention' in the sip phone's context in sip.conf Paul - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 12, 2005 11:13 PM Subject: [Asterisk-Users] Grandstream Bugetone 101 mwi I tried to use message waiting indicator, by Subscribe for MWI in the web menu of the phone. However, it does not light up / flash, even if a voice mail is waiting. Where is the switch to turn it to? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy reliability issues
Hmmm I could certainly see that being the issue. If it is the issue, though, then I think it's something that needs to be addressed. In my opinion, Digium needs to address it, as well as the whole provisioning via cli thing. I know Asterisk itself is a CLI oriented piece of software, but the more one can do do decrease configuration timing and issues the better off one is. I think it would be a benefit to allow the IAXy to be programmed via web interface. For that matter, from what I can tell via my own experimentation, it appears that you cannot use DNS to define the asterisk server to it. This is bad, since it means that if the IP of the asterisk server changes, you need to directly reprovision *all* of your IAXy devices For a new product, it has potential, hopefully these things will be addressed regards Paul - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 30, 2004 6:14 AM Subject: Re: [Asterisk-Users] IAXy reliability issues On Thu, 30 Dec 2004, Gary wrote: On Thu, 30 Dec 2004 00:12:51 -0700, Paul Fielding wrote: I've just picked up a pair of IAXy devices. They work fine except that they keep going offline. As in, I plug it in, it connects to Asterisk, I can dial and phone and all is dandy. Then, maybe 12h later, maybe 24, maybe 36, maybe 48, I'll either try to phone the device and not get through or I'll pick it up and the dialtone is gone. it's simply lost it's connection to Asterisk. If I unplug and plug back in, it reconnects and all is well. I'm running firmware v. 22. Anyone else experiencing this? Paul DHCP timeouts ?? Didn't somebody say that the IAXy doesn't renew its DHCP lease (ie its a BOOTP client). In which case, your DHCP server needs to give it an infinite lease. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy issues
I still am not sure how to check and/or upgrade firmware. So far the only way I've found to upgrade the firmware is to update the Asterisk code to more recent code, which includes newer firmware for the IAXy. The next time the IAXy connect to Asterisk, it will automatically install the new firmware. This is another concern that I have with the IAXy. With the problems I was having, I updated Asterisk to the latest CVS head in order to get newer firmware for the IAXy, hoping it would resolve some of my issues. Not only did it not resolve my issues, but it created new ones in Asterisk, such as the fact that now (Dec 27 Head) Asterisk cores whenever a SIP client tries to register an extention that doesn't exist. There really ought to be a way to update the firmware without updating Asterisk. Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXy reliability issues
I've just picked up a pair of IAXy devices. They work fine except that they keep going offline. As in, I plug it in, it connects to Asterisk, I can dial and phone and all is dandy. Then, maybe 12h later, maybe 24, maybe 36, maybe 48, I'll either try to phone the device and not get through or I'll pick it up and the dialtone is gone. it's simply lost it's connection to Asterisk. If I unplug and plug back in, it reconnects and all is well. I'm running firmware v. 22. Anyone else experiencing this? Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Soft phone vs. Hardware SIP device quality?
I've been messing with some hardware sip devices and with softphones (X-Lite, X-Pro and SjPhone). Compared to the hardware devices, the softphones blow chunks (tm) for sound quality. the softphones are quieter, crackly, and overal more difficult to understand the voice, while a sip device at the same location sounds great. I've tried proprietary softphones (such as Nortel and one other who's name I forget) that sound great, and X-Lite and sjphone are both using the same protocol and codecs as the sip devices, so i don't understand what I'm missing. Anyone have any suggestions? regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Crackly Bad quality
I'm interested in this, too. I find that when I use Xten or SjPhone software locally the quality is quite good, but when I use it remotely across the internet, I get quite a crackly response. *however*, if I use some SIP hardware, such as a Grandstream 236 or an IP phone (still use alaw just like Xten and SJ), the quality is great, even from halfway around the world. Literally. This leads me to think that the softphones are doing something not as well as the hardware SIP devices. Anyone have any thoughts on that? I've seen this behavior with multiple client computers, so I don't think it's just the computer that's using the softphone that's to blame... Paul - Original Message - From: Bruno Hertz [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, December 18, 2004 4:37 PM Subject: Re: [Asterisk-Users] Asterisk Crackly Bad quality On Sat, 2004-12-18 at 14:55 -0600, Steven Critchfield wrote: I highly suggest you work on getting either the RTC or USB driver loaded to provide timing if you don't already have a PSTN card for that job. OK, this is all softphones and one AVM passive BRI card here, so no digium hardware. And frankly, I'd be rather surprised if asterisk, apart from the standard kernel rtc timer, needs a special timer just to play back the demo voice and send it over the LAN. Remember, it's the initial setup we're talking about, and only the demo playback. To make sure, I compiled and loaded the ztdummy driver (from zaptel dir for 2.6 kernel). No difference. Also, if it really was the timer, that would hardly explain why e.g. FC3 and Debian Sarge behave so (wildly) different. I admit though that strange things happen sometimes :) So no, the dummy driver didn't do it. Thanks for your hints, Bruno. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729, x-pro, and codec ordering
Try setting the codec settings for each peer instead of under the general heading. I've tried that - if disallow=all,allow=g729 in the appropriate peer then it will indeed use the correct codec. What I'd like to do, though, is be able to switch codecs depending on my location. Currently the only way I can see to do that is to force the codec within the peer in sip.conf every time I want to change between g729 or ulaw, which means SSHing into my asterisk server to make the change every time. I was hoping there would be a slightly more elegant way to do things... :( Paul On Tuesday 07 December 2004 05:39 am, Paul Fielding wrote: I'm in the middle of getting g729 to work on my server and running into odd stuff. The issue revolves around what appears to be a much talked about (but not seeming to be much solved) issue of selecting which codec gets used at a given time. I have two g729 licenses. I'd like to be able to get asterisk to use g729 (via x-pro) only when I want to, reason being that if I'm in a high bandwidth environment I'd rather have the higher quality of ulaw, but when I'm in a low bandwidth environment I'd like to select g729. There doesn't seem to be much rhyme or reason to which codec gets chosen, and it seems to vary depending on whether the call is outgoing or incoming. And furthermore, turning off a codec in x-pro doesn't seem to do anything. For example, if I have: [general] disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm allow=ilbc and then dial out on x-pro, G729 is selected. Then I turn off G729 and turn on g711u (I make g711u the only black codec on the x-pro display), then make a call, the call is still made using G729. Further more, with the same settings if I call from a zap channel to the x-pro sip extension, the codec chosen is g711u, even though I might only have g729 enabled on x-pro, and even though g729 is the first one on the list above. Anyone have any suggestions, or can point me to something to read? regards, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729, x-pro, and codec ordering
- Original Message - From: Eric Wieling aka ManxPower [EMAIL PROTECTED] You can always allow both codecs and this only allow the codec you want in X-Pro. Asterisk won't try to use ulaw if the phone says it doesn't allow ulaw. what I'd like to do is be able to switch between G729 and ulaw. if I allow only those two codecs, the problem I have is what I mentioned previously (see below), and I can't select which codec I want to use. The goal is to pick the codec depending on whether I'm in a high bandwidth or low bandwidth environment regards, Paul From: Paul Fielding I have two g729 licenses. I'd like to be able to get asterisk to use g729 (via x-pro) only when I want to, reason being that if I'm in a high bandwidth environment I'd rather have the higher quality of ulaw, but when I'm in a low bandwidth environment I'd like to select g729. There doesn't seem to be much rhyme or reason to which codec gets chosen, and it seems to vary depending on whether the call is outgoing or incoming. And furthermore, turning off a codec in x-pro doesn't seem to do anything. For example, if I have: [general] disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm allow=ilbc and then dial out on x-pro, G729 is selected. Then I turn off G729 and turn on g711u (I make g711u the only black codec on the x-pro display), then make a call, the call is still made using G729. Further more, with the same settings if I call from a zap channel to the x-pro sip extension, the codec chosen is g711u, even though I might only have g729 enabled on x-pro, and even though g729 is the first one on the list above. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729, x-pro, and codec ordering
X-Pro does not allow you to only enable one codec? Ostensibly so. I can disable the codecs at any time in X-Pro. Problem is, it doesn't seem to work. I can disable a codec, and then when Asterisk connects, the codec will magically light back up and get used, even though I've disabled it. *shrug*. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729, x-pro, and codec ordering
I'm in the middle of getting g729 to work on my server and running into odd stuff. The issue revolves around what appears to be a much talked about (but not seeming to be much solved) issue of selecting which codec gets used at a given time. I have two g729 licenses. I'd like to be able to get asterisk to use g729 (via x-pro) only when I want to, reason being that if I'm in a high bandwidth environment I'd rather have the higher quality of ulaw, but when I'm in a low bandwidth environment I'd like to select g729. There doesn't seem to be much rhyme or reason to which codec gets chosen, and it seems to vary depending on whether the call is outgoing or incoming. And furthermore, turning off a codec in x-pro doesn't seem to do anything. For example, if I have: [general] disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm allow=ilbc and then dial out on x-pro, G729 is selected. Then I turn off G729 and turn on g711u (I make g711u the only black codec on the x-pro display), then make a call, the call is still made using G729. Further more, with the same settings if I call from a zap channel to the x-pro sip extension, the codec chosen is g711u, even though I might only have g729 enabled on x-pro, and even though g729 is the first one on the list above. Anyone have any suggestions, or can point me to something to read? regards, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] x-lite audio not working correctly (very LOoooow and SLoooow)
I installed x-lite on a new system, and can't make it work. it connects to the asterisk server fine, but it's outgoing audio is messed. Incoming audio is fine and dandy, but outgoing sounds like someone ran it through a voice deepening machine that makes it so low and slow that it's incomprehendable (sp?). The dummy trick of uninstalling-reinstalling doesn't work, and all the settings appear to be identical to another system in which everything is fine. I installed sjphone on the system and it works. I'd rather use x-lite, though, as I don't like sjphone's interface... regards, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using Pocket PC over cell phone connection?
Anyone tried using a pocket pc with sjphone or x-ten over a cell phone connection? I'd like to be able to connect using my cell phone data connection, but so far I've come across bandwidth constraints - The closest to success I've found so far is to use the GSM codec, but even then the bandwidth seems to much for it. Anyone had any luck? Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Top posting
Whatever. I find it frankly more annoying to have people bottom post. I use Outlook Express for my mail (as do millions of others), and the way OE formats it's mail lends itself to top posting.When you bottom post, I need to scroll way down the message to see your response, while when you top post I can see the response right away. If I want to see the source message *then* I'll scroll down, but chances are I've already been reading the thread so this isn't necessary. Professional? That's a matter of opinion, I don't think it's any less professional to top post, it's purely a question of what's convenient for different readers. Besides, as has already been commented on before, people should just be happy that everyone's willing to spend their time offering their advice on this forum rather than being concerned about how their message is formatted... just my 2 cents... Paul - Original Message - From: Tracy R Reed [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 4:15 AM Subject: Re: [Asterisk-Users] Re: Top posting On Fri, Nov 12, 2004 at 06:57:05PM -, Kevin Walsh spake thusly: up properly. There is no excuse at all for lazily top-posting. As a businessman I also see it as a matter of professionalism. I see top posting and not trimming etc as just unprofessional. I regularly do get poorly formatted emails with no trimming and top posting and such emails always strike me as unprofessional and amature. To some degree email is not all that unlike traditional written communications. You would not send a client such a poorly formatted letter. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Top posting
Feel free to debate and argue, but to litter your response with personal insults to me simply tells everyone that your response is worth even less than my measely 2 cents. If you want to make it personal, take it to email rather than this forum so the others don't have to waste their time with it... Sorry everyone, this is the last public comment I'll make on the issue... :( regards, Paul - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 4:45 PM Subject: RE: [Asterisk-Users] Re: Top posting Paul Fielding [EMAIL PROTECTED] lazily top-posted: Whatever. I find it frankly more annoying to have people bottom post. I use Outlook Express for my mail (as do millions of others), and the way OE formats it's mail lends itself to top posting. As you seem to find it difficult to move the cursor on your own, perhaps this utility will help: http://home.in.tum.de/~jain/software/oe-quotefix/ You could install it to fix your broken mail reader - if it's not too much effort. When you bottom post, I need to scroll way down the message to see your response The effort involved is clearly too much for you to handle. Are you really that lazy? If I want to see the source message *then* I'll scroll down, but chances are I've already been reading the thread so this isn't necessary. Your laziness will make life difficult for people who find your followups in a future Google search. Just because you've read the entire thread, doesn't mean that someone else will have done the same next year. Then again, the chance of you posting useful information for someone to find in Google does seem to be a bit remote. just my 2 cents That might be all your time is worth. Others get paid a little more than that. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pressing a key to get out of voicemail?
I've currently got Asterisk configured to take incoming calls and send them directly to my voicemail. I'd prefer to keep this approach rather than sending people to a menu first. What I want to be able to do is have voicemail come up, but if someone presses a key, such as 9 or 8 or perhaps a combo 98 or such, have it break out of voicemail and let me authenticate a password, and upon succeeding let me back into a dialplan so I can dial extensions or another outside line. It appears that there's no way to alter the Voicemail app behavior? So far the only way I've come up with to do this is to cheat. Instead of going straight to voicemail I've set it to play a wav file that Backgrounds "This is my voicemail, leave a message.. yada yada", then sends the call to Voicemail, only my Voicemail unavailable message is an empty wave file. This allows me to press another extension while the first wave file is being played, and as long as I do it before it jumps into voicemail, I can break to another context where I can Authenticate, then send where I want. But this is a kludge, and I cannot change my voicemail message using regular voicemail tools this way. I'd rather set it up properly. Any ideas? regards, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Authenticate or DISA?
I want to authenticate to the phone system, then be able to call an extension or dial an outside line. My preferred method would be to use DISA, because a) it's non-verbal - ie. it doesn't talk, just provides dialtone, and b) it provides dialtone. However, it seems to be unreliable. when I phone in, sometimes it doesn't seem to recognize my DTMF, and just keeps giving a dialtone without authenticating. It's inconsistent. I can phone the system and it'll work, phone again and it doesn't, phone a third time and it works. My alternative seems to be to use Authenticate, and upon authenticating simply send the caller to the appropriate context to punch in extensions or calls. The problem with this is a) it voices the authentication - ie "please enter password" which to me is inviting people to try to figure it out, and b) after authenticating you don't get a dialtone, just silence. But at least it works reliably every time. Any thoughts? regards, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pressing a key to get out of voicemail?
Go figure. I guess I need to use 'show applications' more. I searched all over for docos on the Voicemail app and it was under my nose the whole time... :) Guess I'm on my learning curve. thanks a bunch... Paul - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, November 12, 2004 6:34 PM Subject: Re: [Asterisk-Users] pressing a key to get out of voicemail? Paul Fielding wrote: I've currently got Asterisk configured to take incoming calls and send them directly to my voicemail. I'd prefer to keep this approach rather than sending people to a menu first. What I want to be able to do is have voicemail come up, but if someone presses a key, such as 9 or 8 or perhaps a combo 98 or such, have it break out of voicemail and let me authenticate a password, and upon succeeding let me back into a dialplan so I can dial extensions or another outside line. This is from show application voicemail If the caller presses '0' (zero) during the prompt, the call jumps to extension 'o' in the current context. If the caller presses '*' during the prompt, the call jumps to extension 'a' in the current context. Of course if you press # it will exit out of voicemail to. Voicemail will even TELL you to press # after leacing a message. Once voicemail exits, of course the dialplan will continue at the next priority. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling an outside number along side other internal extensions?
I've currently configured incoming calls to simultaneously ring an analogphone (via TDM400P) and two SIP phones. I'd like to have it also simultaneously dial out the TDM400P on a PSTN to ring my cell phone, and have the first one to answer win the battle. In my digging I've figured out that I can add the Zap channel to the dial list, such as Dial(SIP/7001SIP/7002ZAP/3/5551212,20), however when I include the PSTN line in this command (ZAP/3/) I get an interesting thing happening. All SIP phones start ringing. Asterisk connects ZAP/3 to dial out and dials out Asterisk then says to the effect of "ZAP/3 has answered the call" (since the line has now gone off hook) and stops ringing all the SIP phones immediately, leaving me with only the cell phone ringing. It then fails to go to Voicemail and just keeps ringing the cell phone, because as far as Asterisk is concerned the line has been bridged and is connected. Any suggestions? regards, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling an outside number along side otherinternal extensions?
Hmmm... Interesting that you mention it's not a problem with VOIP companies as they use PRI. The analog trunk I'm connecting to is actually a Vonage line. Thing is, it seems to me that by connecting via the Zap channel to the Vonage ATA I'm effectively cancelling any advantage that Vonage's PRI might have... (?).I don't believe I have any other alternatives for connecting to Vonage's service, but perhaps I'm wrong about that. Perhaps I'll give the c option a try. It looks like it might do the job... regards, Paul - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, November 12, 2004 6:56 PM Subject: Re: [Asterisk-Users] Calling an outside number along side otherinternal extensions? Paul Fielding wrote: I've currently configured incoming calls to simultaneously ring an analog phone (via TDM400P) and two SIP phones. I'd like to have it also simultaneously dial out the TDM400P on a PSTN to ring my cell phone, and have the first one to answer win the battle. In my digging I've figured out that I can add the Zap channel to the dial list, such as Dial(SIP/7001SIP/7002ZAP/3/5551212,20), however when I include the PSTN line in this command (ZAP/3/) I get an interesting thing happening. All SIP phones start ringing. Asterisk connects ZAP/3 to dial out and dials out Asterisk then says to the effect of ZAP/3 has answered the call (since the line has now gone off hook) and stops ringing all the SIP phones immediately, leaving me with only the cell phone ringing. It then fails to go to Voicemail and just keeps ringing the cell phone, because as far as Asterisk is concerned the line has been bridged and is connected. Any suggestions? Analog FXO ports ae considered answered as soon as the dialing is finished. Nothing you can do about this because there is no way for Asterisk to know when the far end answers. This is not a problem with (most) Channelized Voice T-1, it's not a problem with PRI and not a problem with VoIP telephone companies, since they all use PRI. You can sort of work around this problem by using the poorly documented c option to the Zap dial command. Something like Zap/1c or something like that. I've never used it. That option requires the callee press # to accept the call. No sound file is played. See the mailing list archives. It's been discussed off and on. --Eric --Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400p module error?
I've managed to get my Asterisk server up and working, and my TDM400p seems to be working fine, inside and outside lines. I was pouring through the syslog looking for a different issue when I noticed the following 'Prosilc Failed' message (9 lines in): Nov 11 17:34:36 natasha kernel: Zapata Telephony Interface Registered on major 196Nov 11 17:34:36 natasha kernel: PCI: Found IRQ 10 for device 00:0b.0Nov 11 17:34:36 natasha kernel: PCI: Sharing IRQ 10 with 00:11.0Nov 11 17:34:36 natasha kernel: Freshmaker version: 71Nov 11 17:34:36 natasha kernel: Freshmaker passed register testNov 11 17:34:36 natasha kernel: Module 0: Installed -- AUTO FXS/DPONov 11 17:34:36 natasha kernel: Timeout waiting for calibration of module 1Nov 11 17:34:36 natasha kernel: Timeout waiting for calibration of module 1Nov 11 17:34:36 natasha kernel: Proslic Failed on Second Attempt to Auto CalibrateNov 11 17:34:36 natasha kernel: Module 1: Installed -- MANUAL FXSNov 11 17:34:36 natasha kernel: Module 2: Installed -- AUTO FXO (FCC mode)Nov 11 17:34:36 natasha kernel: Module 3: Installed -- AUTO FXO (FCC mode)Nov 11 17:34:36 natasha kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)Nov 11 17:34:36 natasha kernel: Registered tone zone 0 (United States / North America) Should I be concerned, or is this normal? regards, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
