[asterisk-users] Too Few Fax Detections
OK, after my last message about fax detection, I feel a bit better informed and able to press forward. I started looking into this because I was getting lots of false positive fax detection errors in the logs with faxdetect=both set in chan_dahdi.conf. Anyhow, I do not currently use fax detection, and we have a dedicated Fax DID on our PRI, so setting faxdetect=no works fine. Having said that, I would like to sort it out as I may want to use fax detection in the future. Unfortunately, I seem to be having odd results. I set faxdetect=incoming last night and restarted dahdi and asterisk. Since that time, we have received 17 faxes, but I only have three fax detections in my asterisk log, so far as I can tell: # grep -i fax /var/log/asterisk/full [Jan 5 05:53:39] NOTICE[6686] chan_dahdi.c: Fax detected, but no fax extension [Jan 5 10:24:27] NOTICE[11834] chan_dahdi.c: Fax detected, but no fax extension [Jan 5 11:48:52] NOTICE[13804] chan_dahdi.c: Fax detected, but no fax extension All three calls listed are indeed fax calls, and since there is no fax extension in that context, the call just proceeds along as if nothing happened (which is appropriate). My question is this: If I have received 17 faxes since enabling fax detection, shouldn't I see ~17 entries in the log? Assuming the answer to that question is yes, what might be causing the system to not detect faxes on the other 14 calls? Many thanks, Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM410 and DSL
Hi all, I have a system installation in Guam with two trunks. One has a DSL service riding on it with the usual filter. That channel however keeps throwing alarms. I bypassed the filter and it stopped throwing alarms, but of course the high frequencies annoy the users. I swapped the filters and the alarms came back. Any suggestions? Could I have a bad DSL modem? Cassius -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] using google for vm transcripts
I'm pretty impressed by how well (comparatively) google voice does in
doing voice mail transcripts. So I'd like to have google do my local
voice mail, and then email the transcript.
So I set up extensions.conf:
exten =s,n,Dial(${House_Phones},36) ; this should be six rings
exten =s,n,Dial(Gtalk/my-user-name/${my-gv-number{...@voice.google.com)
but I get this error:
-- Executing [...@incoming-pstn-line:6] Dial(DAHDI/4-1,
Gtalk/my-user-name/my-gv-number@voice.google.com) in new stack
[Jan 5 16:26:22] ERROR[3129]: chan_gtalk.c:1871 gtalk_request: No XMPP
client to talk to, us (partial JID) : my-user-name
Any help appreciated.
sean
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Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)
If you do get a Polycom, the old 501 (discontinued) have a louder ring (or can be configured to have a louder ring, don`t quite remember) then the newer ones. But the others are right: it's not meant for this, at least not in a noisy environment. What can work though is a Polycom 321, with a (loud) speaker plugged into the 3.5mm port and properly configured to have the speaker take the call (see paging app and Polycom admin manual). It`s a bit of a hassle but it`s much better than the unreliable and expensive Cyberdata paging products (I hated the one I tried, replaced it with a 321 as described). Mike Ah.. so you've used the Cyberdata intercom and didn't like it. What about it was unreliable? Thank you for the inp. ut. Not the intercom, the paging server. It was in a very active environment (car dealership, sometimes many pages per minute). It just stopped responding for a few minutes once in a while. The config is actually very easy. Under less load, it worked well. What loud speaker did you end up going with? Polycom 321 with a 3.5mm plug to an external speaker. They already had something in place speaker-wise, so didn't bother checking. Was it cumbersome (space-wise) to have a phone and a loudspeaker? Space wasn't an issue there, it was a Polycom 321 connected to a building-wide paging system in a server room. I imagine on a busy kitchen wall it's different. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dtmf-troubles with Snom
Hello list, I'm having DTMF-troubles with a Snom phone. I want to know if it's the Snom or Asterisk that makes the trouble. I'm playing a prompt, then make a choice for 2 : [Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] -- SIP/test1-0701 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' (language 'nl') [Jan 5 17:06:39] VERBOSE[29172] pbx.c: [Jan 5 17:06:39] -- Executing [...@sub-routing:52] WaitExten(SIP/test1-0701, 15) in new stack *[Jan 5 17:06:41] DTMF[29172] channel.c: DTMF begin '2' received on SIP/test1-0701 [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF begin ignored '2' on SIP/test1-0701 [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on SIP/test1-0701, duration 160 ms [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on SIP/test1-0701* What follows is a prompt again, and it automatically chooses option 2 : [Jan 5 17:06:41] VERBOSE[29172] file.c: [Jan 5 17:06:41] -- SIP/test1-0701 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl') *[Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on SIP/test1-0701, duration 160 ms [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on SIP/test1-0701* Even without pressing 2 on the Snom phone, option 2 is chosen in the menu. The above is different when I do the same with a Grandstream device : [Jan 5 17:14:15] VERBOSE[29384] file.c: [Jan 5 17:14:15] -- SIP/test6-0714 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' ( language 'nl') [Jan 5 17:14:17] VERBOSE[29384] pbx.c: [Jan 5 17:14:17] -- Executing [...@sub-routing:52] WaitExten(SIP/test6-0714, 15) in new stack [Jan 5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan 5 17:14:18] doing dnsmgr_lookup for 'ssw4.brussels.weepee.org' [Jan 5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan 5 17:14:18] doing dnsmgr_lookup for 'ssw4.brussels.weepee.org' *[Jan 5 17:14:21] DTMF[29384] channel.c: DTMF begin '2' received on SIP/test6-0714 [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF begin ignored '2' on SIP/test6-0714 [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF end '2' received on SIP/test6-0714, duration 100 ms [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF end passthrough '2' on SIP/test6-0714* [Jan 5 17:14:38] VERBOSE[29384] file.c: [Jan 5 17:14:38] -- SIP/test6-0714 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl') [Jan 5 17:14:39] VERBOSE[29384] pbx.c: [Jan 5 17:14:39] -- Executing [...@sub-routing:52] WaitExten(SIP/test6-0714, 15) in new stack *[Jan 5 17:14:44] DTMF[29384] channel.c: DTMF begin '2' received on SIP/test6-0714 [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF begin ignored '2' on SIP/test6-0714 [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF end '2' received on SIP/test6-0714, duration 100 ms [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF end passthrough '2' on SIP/test6-0714* Here I explicitly chose option 2 by pressing on button 2. What is going on with the Snom ? There is a difference in duration (160ms vs 100ms). Is that the problem ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cannot answer incoming calls
Have recently installed some Snom phones into an office. Phones are natted and connect to a 1.4 server on a public IP We can make outgoing calls, but are unable to answer incoming calls. The phone rings, but the call cannot be picked up. Other phones on other sites connected to the server are working perfectly. Looking at the SIP trace it appears the phone transmits: Sent to udp:193.33.xx.xx:5060 at 6/1/2011 11:49:20:868 (849 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 193.33.xx.xx:5060;branch=z9hG4bK6e82052c;rport=5060 From: xx sip:07765000...@sip3.office-voip.com;tag=as1b6fc27c To: sip:x_...@79.123.xx.xx:25380;tag=37gg1zu3wp Call-ID: 1b212085091e98387237125f0ab81...@sip3.office-voip.com CSeq: 102 INVITE Contact: sip:x_...@192.168.4.19:2048;reg-id=1 User-Agent: snom300/7.3.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 220 v=0 o=root 641540583 641540584 IN IP4 192.168.4.19 s=call c=IN IP4 192.168.4.19 t=0 0 m=audio 52386 RTP/AVP 18 101 a=rtpmap:18 g729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv but it is never received by the server. Interestingly RINGING and REGISTER messages are working OK. The NAT router is out of our control. Are we looking at a SIP ALG getting in the way? Thanks, John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Log and forward calls to cellphone?
On Tue, 04 Jan 2011 17:57:27 +, Sebastian s...@open-t.co.uk wrote: Sorry to keep on butting in. I've been interested in SIP on Android for a while now - so this just gave me more incentives to actually do the research :-) No problem. I hadn't thought about using a 3G connection to register a smartphone with Asterisk and receive calls directly that way. Thanks for the tip. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Log and forward calls to cellphone?
On Wed, 05 Jan 2011 11:49:40 +0100, Administrator TOOTAI ad...@tootai.net wrote: As I told, the best SIP client I had is Nokias one. Fully integrated, working out of the box. Thanks much for the feedback. I was mentioning OpenVPN because I assumed 3G carriers blocked SIP, but your experience shows that they don't necessarily do. I'll check the Nokia E series and the latest Android phones. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind Transfer not working - 1.4.38
On Wed, 2011-01-05 at 15:47 +, Ishfaq Malik wrote:
Hi
We've been running asterisk 1.4.17 (deb package) in a production
environment for some while now and are finally taken the plunge to
update to 1.4.38 (Ubuntu servers). All of this is using the RealTime
Architecture
I have upgraded the asterisk version in one of our test environments and
blind transferring seems to have suddenly stopped working. It was
working fine under 1.4.17
So, call comes in to extension 501 who does a blind transfer to
extension 504 at which point the call gets completely cut off.
I ran a SIP trace of this happening and it appears to be attempting to
do the transfer:
-
--- (12 headers 0 lines) ---
Call 7c5d5a603b2803fd7e451de82...@x.x.x.x got a SIP call transfer from
caller: (REFER)!
SIP transfer to extension 5...@pack-local by pack...@domain.co.uk
--- Transmitting (NAT) to x.x.x.x:52753 ---
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP
192.168.1.105:3072;branch=z9hG4bK-sgoqylu125ma;received=x.x.x.x;rport=52753
From: sip:pack...@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
To: incoming mobile number sip:incoming mobile
number@x.x.x.x;tag=as4d0dbc04
Call-ID: 7c5d5a603b2803fd7e451de82...@x.x.x.x
CSeq: 2 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:incoming mobile number@x.x.x.x
Content-Length: 0
set_destination: Parsing sip:pack...@192.168.1.105:3072;line=guuuyf05 for
address/port to send to
set_destination: set destination to 192.168.1.105, port 3072
Reliably Transmitting (NAT) to x.x.x.x:52753:
NOTIFY sip:pack...@192.168.1.105:3072;line=guuuyf05 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK121bb8ff;rport
From: incoming mobile number sip:incoming mobile
number@x.x.x.x;tag=as4d0dbc04
To: sip:pack...@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
Contact: sip:incoming mobile number@x.x.x.x
Call-ID: 7c5d5a603b2803fd7e451de82...@87.237.58.231
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: incoming mobile number sip:incoming mobile
number@x.x.x.x;privacy=off;screen=no
Event: refer;id=2
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 21
SIP/2.0 183 Ringing
___
But as stated above, extension 504 doesn't ring and the call dies.
Now 504 is a valid extensions in the context pack-local
select * from extensions where exten='_5XX';
+---++---+--+---+---+
| id| context| exten | priority | app | appdata
|
+---++---+--+---+---+
| 65127 | pack-local | _5XX |1 | Macro |
stdexten|${EXTEN}|pack-local|PACK |
+---++---+--+---+---+
Also, attended transfers work without a problem.
Both SIP phones used were Snom phones.
Has anyone encountered an issue like this before?
I spotted something new here, when I try to do the blind transfer I get
the following output on the console
== Spawn extension (pack-local, 504, 0) exited non-zero on
So why would it be looking at priority 0 rather than priority 1?
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] Too Few Fax Detections
On 01/05/2011 08:12 PM, Thomas Rymes wrote: OK, after my last message about fax detection, I feel a bit better informed and able to press forward. I started looking into this because I was getting lots of false positive fax detection errors in the logs with faxdetect=both set in chan_dahdi.conf. Anyhow, I do not currently use fax detection, and we have a dedicated Fax DID on our PRI, so setting faxdetect=no works fine. Having said that, I would like to sort it out as I may want to use fax detection in the future. Unfortunately, I seem to be having odd results. I set faxdetect=incoming last night and restarted dahdi and asterisk. Since that time, we have received 17 faxes, but I only have three fax detections in my asterisk log, so far as I can tell: # grep -i fax /var/log/asterisk/full [Jan 5 05:53:39] NOTICE[6686] chan_dahdi.c: Fax detected, but no fax extension [Jan 5 10:24:27] NOTICE[11834] chan_dahdi.c: Fax detected, but no fax extension [Jan 5 11:48:52] NOTICE[13804] chan_dahdi.c: Fax detected, but no fax extension All three calls listed are indeed fax calls, and since there is no fax extension in that context, the call just proceeds along as if nothing happened (which is appropriate). My question is this: If I have received 17 faxes since enabling fax detection, shouldn't I see ~17 entries in the log? How are you delivering the inbound FAX calls to your FAX machine? If you are sending them back out a DAHDI channel (to an FXS port on an analog card, for example), then as soon as the two channels are bridged the audio never comes up to Asterisk (under normal circumstances), it stays in DAHDI, so the Asterisk DSP can't detect the CNG tone. If the FAX machine answers the incoming call fairly quickly, there may not be any opportunity for the CNG to be detected. In addition, you may not be even receiving any audio from the calling FAX machine until you answer the incoming channel (depending on your PRI provider). If you want to have the best chance to detect each incoming FAX using the Asterisk DSP, you'll have to answer the incoming channel as soon as it hits the dialplan, then wait 3 or 4 seconds, then send the call onwards to your actual FAX machine. FAX detection is really expected to be used on calls that would otherwise be answered by a non-FAX endpoint (IVR, voicemail, user with a phone, etc.) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Benefit of PRI vs SIP trunk calls
Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a PRI line compared to a call via a SIP trunk? As an example, in a PRI call there is this message that shows up on the console: [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network. for a call to a fax machine. Does asterisk set anything that a dialplan can access that can know the call was to a fax machine? If a call is placed to a number that is disconnected so a special information tone is played can either a PRI call or a SIP call know this without analyzing the audio stream? Are there reasons to prefer the use of PRI over SIP or SIP over PRI? I would like people's opinions as to if one form is better than the other in any meaningful way. Thanks for you feed-back. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)
On 01/05/2011 01:51 PM, Tom Rymes wrote: On 01/05/2011 7:50 AM, Andy Graybeal wrote: We've got two noisy kitchens that need to talk back and forth. Andy, Why, exactly, are you trying to combine an inter-kitchen intercom and your phone system? Might it make more sense to have a non-phone-based intercom system, plus a phone for making phone calls? Tom Tom, Good question. I'm not sure, but maybe I was hoping to kill two birds with one stone. I will take your suggestion into account as I'm not sure what to do. Do you have any intercom system recommendations? Would it be POE also, and something I could manage with Asterisk? -Andy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird phone behavior after recent CentOS 5 update
On Wed, Jan 5, 2011 at 6:59 PM, Myles Wakeham my...@techsol.org wrote: For some reason our Asterisk box is doing something really unusual following applying a routine update to CentOS 5 on Monday. We have Asterisk 1.4.2 and its been working great for years. But now when the phone system receives an incoming SIP call, its not providing any audible dial sound to any caller. It is recognizing the incoming call, and after no answer for about 5 rings or so, it goes to voice mail. But there is no audible 'ring' to the caller. Just nothing - blank, empty silence. Of course any automated answering system (ie. business phone menu, etc.) that we have works just fine. Its just the lines that go directly to an internal phone that are no longer providing any audible ring which is sending a message to the caller that their call didn't go through. Does anyone have any idea what might cause this? Definitely check your firewall settings. Definitely consider rebuilding against the libs as they now exist on your machine. CentOS (which are really RedHat) library changes aren't always fully disclosing the things that actually change. The worst example in recent memory was when a CentOS update changed the defaults to sudo, and you had to go manually override to allow sudo to work without a tty. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On 01/06/2011 05:25 AM, Tim Panton wrote: On 5 Jan 2011, at 13:07, Steve Underwood wrote: G.722.1 is a 7kHz bandwidth codec. G.722.1C is a stretched version offering 14kHz bandwidth. These are most often found in Polycom phones, but they are available elsewhere. The only widely supported HD codec is G.722. Pretty much anything offering wideband voice supports G.722. Except skype which only supports SiLK as the HD codec. I mention this because most people's experience with HD will be in a Skype-to-skype call, although admittedly not in this group. That's a very good point, although Skype does support more codecs than just Silk, and I believe G.722 may be one of them. Nonetheless, it is Silk that people have got used to. It offers about 11kHz bandwidth, so it is wider band than G.722. The critical addition than wideband gives over normal telephony is the 5kHz to 7kHz area, where a lot of the energy that allows us to differentiate the unvoiced phoneme lies. The energy between 7kHz to 15kHz does, however, add a lot to the human voice, and allows for a more relaxed listening experience - its just less tiring to listen to. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
We should also be very clear that the Siren codecs are supported on the Polycom SoundStation conference phones and the VVX-1500 Business Media Phones. These codecs are not supported in the SoundPoint desk phones. The SoundPoint series support the more basic G.722 codec in the IP335/450/550/560/650/670 models. Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgra...@mstvp.onsip.com skype mjgraves Original Message Subject: Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs? From: Steve Underwood ste...@coppice.org Date: Wed, January 05, 2011 6:09 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com On 01/06/2011 12:05 AM, Kevin P. Fleming wrote: On 01/05/2011 07:07 AM, Steve Underwood wrote: On 01/05/2011 03:29 PM, Bruce B wrote: Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? The G.722 codec in * is G.722. The Siren7 codec in * is probably not Siren 7, but G.722.1. G.722.1 is very similar to Siren7, but uses a different code in the SDP and has some minor differences in the codec. The name G.722.1 may look similar to G.722, but the codecs bear no relation to each other. The Siren14 codec in * is probably not Siren14, but G.722.1C. G.722.1C is very similar to Siren14, but like Siren7/G.722.1 the SDP code is different, and there are minor differences in the codec. Asterisk actually supports both the Siren* and G.722.1* names in SDP negotiations. I wasn't aware there were bitstream incompatibilities between the Siren* and G.722.1* variants, even though the code may be slightly different... so Asterisk uses a single codec module for both variants. I am unclear how compatible or incompatible the bitstreams may be. What I know (from implementing these codecs) is that the source code Polycom provide licencees, as the basis for developing their own G.722.1 and G.722.1C codecs, has several comments referring to things not being quite the same as Siren7/Siren14. However, they don't hand out the actual Siren7/Siren14 source code, so I don't know how much divergence there is. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk replying to wrong port for NOTIFY messages
On Wed, 5 Jan 2011, James Lamanna wrote: See the following SIP trace. Where in the world does Asterisk get port 1025 to respond to? This is asterisk 1.6.x. Hi James, I'm sure it would be the NAT translated port on the public side of the customer's firewall... j Thanks. -- James --- SIP read from zzz.zzz.zzz.44:9363 --- NOTIFY sip:pbx1.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M From: xxx-xxx- sip:xxx...@pbx1.mydomain.com;tag=467525dd6fac949do0^M To: sip:pbx1.mydomain.com^M Call-ID: 707176dd-38f47...@192.168.1.140^m CSeq: 118907 NOTIFY^M Max-Forwards: 70^M Contact: xxx-xxx- sip:xx...@192.168.1.140:9363^M Event: keep-alive^M User-Agent: Cisco/SPA509G-7.4.6-0002fdff90a4^M Content-Length: 0^M ^M - [Jan 5 13:46:36] VERBOSE[3919] logger.c: --- (11 headers 0 lines) --- [Jan 5 13:46:36] VERBOSE[3919] logger.c: --- Transmitting (no NAT) to zzz.zzz.zzz.44:1025 --- SIP/2.0 200 OK^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3;received=zzz.zzz.zzz.44^M From: xxx-xxx- sip:xx...@pbx1.mydomain.com;tag=467525dd6fac949do0^M To: sip:pbx1.mydomain.com;tag=as0493c604^M Call-ID: 707176dd-38f47...@192.168.1.140^m CSeq: 118907 NOTIFY^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO^M Supported: replaces^M Content-Length: 0^M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
