Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip> dtmf-relay rtp-nte codec g711alaw no vad ! When I try to make a call, cisco shows codec g711alaw, but asterisk shows codec g729A (i have the licenses) and there is no audio. When I try disallow=g729, the same occurs, but this time asterisk shows codec gsm. The only way to make a call is allowing only alaw. But this is not convenience, since i need to use g279 with another endpoint (working ok). Why this negotiation problem happens? Thanks Eduardo
I'm working with DIA096B on two remote computers that are behind NAT. They
register ok.
The * has a static public IP address.
I saw other simliar posts btu this seems to be different.
The call is from test2 --> test3:
-- Accepting AUTHENTICATED call from xx.xx.xx.xx , requested format = 2,
actual format = 2
-- Executing Dial("IAX2[test3@test3]/3", "IAX/test2") in
new stack
NOTICE[1200884528]: File app_dial.c, Line 506 (dial_exec): Unable to create
channel of type 'IAX'
== Everyone is busy at this time
-- Executing VoiceMail("IAX2[test3@test3]/3", "u2002")
in new stack
xx.xx.xx.xx is the public ip of the NAT in front of DIAX096B (test2)
[general]
port=5036
bindaddr=publicipaddress
disallow=all
allow=gsm
jitterbuffer=3
tos=reliability
[test2]
type=friend
username=test2
secret=........
host=dynamic
context=test
[tito3]
type=friend
username=test3
secret=......
host=dynamic
context=test
Thanks.
HQ.
----- Original Message ----- From: "Eduardo Goncalves" <eduardo@acenet.com.br> To: <asterisk-users@lists.digium.com> Sent: Tuesday, December 16, 2003 1:08 PM Subject: [Asterisk-Users] codec negotiation> Hi list, > > I'm with a little problem on codec negotiation between a cisco827 and > asterisk. > > My sip.conf is like that: > > [general] > port = 5060 > bindaddr = 0.0.0.0 > context = default > amaflags = default > allow=g729 > allow=gsm > allow=alaw > allow=ulaw > ;disallow=all > > and cisco like that: > > dial-peer voice 6 voip > destination-pattern 0T > session protocol sipv2 > session target ipv4:<asterisk-ip> > dtmf-relay rtp-nte > codec g711alaw > no vad > ! > > When I try to make a call, cisco shows codec g711alaw, but asterisk > shows codec g729A (i have the licenses) and there is no audio. When I > try disallow=g729, the same occurs, but this time asterisk shows codec > gsm. > > The only way to make a call is allowing only alaw. But this is not > convenience, since i need to use g279 with another endpoint (working > ok). >You could try setting the codec before dialing that particular provider. Except I don't see the command now that I'm trying to find it...> Why this negotiation problem happens?Can't help on that one...> > Thanks > Eduardo----- Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise.
We have people connecting to an asterisk box over the internet. They're using the x-lite client behind linksys firewalls. The X-Lite client discovers the firewall no problem and connects to Asterisk without a problem. After connecting the agent shows up properly in "sip show peers" with the IP address of their firewall, etc. They can receive calls no problem. After some time goes by... they don't show as registered with * any more in the sip show peers. They can still make outbound calls, but can not receive the inbound ones. Anyone have any ideas on this one? Thanks, Jonathan
Chris Albertson wrote:> My Strandstream BT100 is working OK for both inbound and outbound now > except that when you speak into the handset you cannot hear your > own voice in the earpeice. It works OK, the other end can hear the > call but most telephone users have become used to hearing their own > voice. > > Is this something I can fix or is it a "feature" of the GS phone?"Sidetone" (hearing yourself talk) should be generated by the phone. I suspect that if it is not there, then the phone is defective and should be replaced. It is not something that is configurable, as far as I know. I have 20 of them on site and they all generate proper sidetone out of the box. The only other possibility I can think of is that silence supression (a GS option) is enabled AND also not working properly. I have never experimented with silence supression. Anyone wand to pipe in? Stephen R. Besch
At my home office I have a X100P card in a server that I've been using for testing. The machine it is in is connected to a HP fax machine and then to the wall outlet. This morning the SBC installer showed up at my house for the ADSL install on that line. He said they detected a short. So he tested the outside box and it was fine. He said it was inside. So we came inside and tested the two devices with his little box. The fax was fine... the X100P card however was causing the short. Now of course I'm going to install a filter on this line for the ADSL, but is this short normal? The installer says that it will kill the ADSL signal. Maybe the X100P is defective? It's been working fine though for making and answering calls to this point. Thanks, Jonathan
Johannes von Drachenfels
2004-Feb-09 10:32 UTC
[Asterisk-Users] no extension in callerid of outgoing calls ...
Hi,
i'm here in germany still fighting against my problems ...
We have a e100p which is sending out his callerid as 78107-0. But what i
need is to send out the extension of the inside callers to, for example:
78107-14
So what i tried is:
exten => _00XX.,1,SetCallerID(78107${CALLERIDNUM})
exten => _00XX.,2,Dial,Zap/g1/${EXTEN:2}
And this is what my asterisk ist telling me:
-- Executing SetCallerID("SIP/14-9707", "7810714") in
new stack
-- Executing Dial("SIP/14-9707", "Zap/g1/1716710815") in
new stack
-- Called g1/1716710815
-- Zap/1-1 is ringing
-- Hungup 'Zap/1-1'
But i still can see only the 78107-0 when i call my mobile ...
Any help would be great ...
Thanks, Johannes
I have outgoing connection to iaxtel and another iax server A. iax server A only accept g729 codec while iaxtel is something I am not quite sure of. At the moment iaxtel only accepts gsm. I remember previously it does accept g729. my problem due to the switching between codec when making outgoing calls to these servers. my iax.conf has these lines: [general] disallow=all allow=gsm allow=g729 I believe the general context define the codec to be used when making outgoing calls. The peer context below general context is to governed codec to be used for incoming calls. Is this correct? now if I specificly disallow g729 in the general context I can make calls to iaxtel. however i cannot make calls to server A as it only accepts g729. After I allow g729, I can make call to server A but the call made to iaxtel cannot go through. The console indicates that the call is accepted by iaxtel using codec 729A, then it says the circuit is too busy. Is there a clever way of governing the codec use for each outgoing connection in order to avoid the issue in codec negotiation? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002
Why do you need 729? I just called your IAXTel number using GSM and connected fine. Michael On Wed, 18 Feb 2004 08:29:48 +0100, dkwok wrote:>I have outgoing connection to iaxtel and another iax server A. > >iax server A only accept g729 codec while iaxtel is something I am not >quite sure of. At the moment iaxtel only accepts gsm. I remember >previously it does accept g729. > >my problem due to the switching between codec when making outgoing calls >to these servers. > >my iax.conf has these lines: > >[general] > >disallow=all >allow=gsm >allow=g729 > >I believe the general context define the codec to be used when making >outgoing calls. The peer context below general context is to governed >codec to be used for incoming calls. Is this correct? > >now if I specificly disallow g729 in the general context I can make >calls to iaxtel. however i cannot make calls to server A as it only >accepts g729. After I allow g729, I can make call to server A but the >call made to iaxtel cannot go through. > >The console indicates that the call is accepted by iaxtel using codec >729A, then it says the circuit is too busy. > >Is there a clever way of governing the codec use for each outgoing >connection in order to avoid the issue in codec negotiation? > >-- >David Kwok > >Iaxtel/FWD # 17001813482 ext 1002 > >_______________________________________________ >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 >-- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com "I believe there comes a time when everything just falls in line." - Pat Benatar, from All Fired Up. ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704
It seems that older version of asterisk does the codec negotiation fine. I have one machine running CVS-12/19/03 and this can negotiate codec g729 and gsm fine. The newer version cvs-1/27/04 does not negotiate codec correctly. The ougoing connection can only go either g729 or gsm. -- David Kwok FWD#/IAXTEL# : 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040221/aeb42bd6/smime.bin
Hello!
I would like to know wether it is possible to have end-to-end codec
negotiation in iax2?
What I mean is...
In case the user dials a number available through PSTN, let's force to
use alaw (the client is in LAN) to overcome unneeded transcoding:
iaxphone->1st asterisk -> PSTN
In case the same user dials a number available throug a chain of IAX2
peers (e.g. 2 peers), try to negotiate the codec end-to-end to consume
less resources for transcoding on asterisk servers (of course, in that
case we don't want to use g711, but ilbc, speex or gsm).
iaxphone->1st asterisk->2nd asterisk->PSTN
Or maybe:
iaxphone->1st asterisk->2nd asterisk->iaxpohone
Is there a way to do that? If yes, how?
Thanks in advance,
TamasJ
Hello Eduardo, Wednesday, December 17, 2003, 1:08:00 AM, you wrote: EG> Hi list, EG> I'm with a little problem on codec negotiation between a cisco827 and EG> asterisk. EG> My sip.conf is like that: EG> [general] EG> port = 5060 EG> bindaddr = 0.0.0.0 EG> context = default EG> amaflags = default EG> allow=g729 EG> allow=gsm EG> allow=alaw EG> allow=ulaw EG> ;disallow=all EG> and cisco like that: EG> dial-peer voice 6 voip EG> destination-pattern 0T EG> session protocol sipv2 EG> session target ipv4:<asterisk-ip> EG> dtmf-relay rtp-nte EG> codec g711alaw EG> no vad EG> ! EG> When I try to make a call, cisco shows codec g711alaw, but asterisk EG> shows codec g729A (i have the licenses) and there is no audio. When I EG> try disallow=g729, the same occurs, but this time asterisk shows codec EG> gsm. EG> The only way to make a call is allowing only alaw. But this is not EG> convenience, since i need to use g279 with another endpoint (working EG> ok). EG> Why this negotiation problem happens? Try to add to cisco peer (not shown in your mail) [cisco] .... disallow=all allow=alaw -- Best regards, Nguyen mailto:dtkhang@hn.vnn.vn
Hello On every Incoming SIP and IAX call I see the following in asterisk debug: Accepting AUTHENTICATED call from 10.10.10.10, requested format = gsm, requested prefs = (), actual format = g729, my prefs (g729|gsm|g723|g726|ulaw|alaw) priority = mine The problem is that the codec preference on both parties is different The calling party has preference gsm/g729/etc The called party (the one you see this debug from) has preference g729/gsm/etc The problem is.. This call is now set up with G729... And I want it rather to be decided by the callING party (thus want the call to be negotiated GSM) What can I do about this? (I just want that if I receive a call the calling party decides the codec, and not my side) My IAX.conf and SIP.conf have the following allow settings now Allow=all Allow=g729 Allow=gsm Allow=ulaw Allow=alaw Help :-)
I don't want that... because - for outbound calls I want priority to be g729 first - for inbound calls I want no priority at all (e.g. the calling asterisk to decide which codec we will use) The last doesn't happen.. This by the way DID happen correctly with previous versions of asterisk (1.0.3 for example) the current CVS-HEAD version doesn't -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mohammed Salim Sent: dinsdag 25 januari 2005 22:10 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Codec negotiation The order matters in asterisk so if you want GSM to take priority over G729, simply put that ahead of the G729... so your settings should be: Allow=all Allow=gsm Allow=g729 Allow=ulaw Allow=alaw Try that and see if it works. Regards, Mohammed Salim EZZI Telecom, Inc. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mark Eissler Sent: Tuesday, January 25, 2005 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Codec negotiation The codec is selected by asterisk depending upon the codecs that you have allowed for the particular channel context and your setting of the bandwidth= parameter. It would be nice if you could set things up so that an inbound call could force * to a higher bandwidth codec when needed (for example, an inbound fax call, let's say) but AFAIK this is not possible. -mark On Jan 25, 2005, at 10:18 AM, <niels@wxn.nl> wrote:> > Hello > > On every Incoming SIP and IAX call I see the following in asterisk > debug: > > Accepting AUTHENTICATED call from 10.10.10.10, requested format = gsm, > requested prefs = (), actual format = g729, my prefs > (g729|gsm|g723|g726|ulaw|alaw) priority = mine > > The problem is that the codec preference on both parties is different > > The calling party has preference gsm/g729/etc > The called party (the one you see this debug from) has preference > g729/gsm/etc > > The problem is.. This call is now set up with G729... And I want it > rather to be decided by the callING party (thus want the call to be > negotiated GSM) > > What can I do about this? (I just want that if I receive a call the > calling party decides the codec, and not my side) > > My IAX.conf and SIP.conf have the following allow settings now > > Allow=all > Allow=g729 > Allow=gsm > Allow=ulaw > Allow=alaw > > > Help :-) > > > _______________________________________________ > 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 >-- Mark Eissler, mark@mixtur.com Mixtur Interactive, Inc. -@- http://www.mixtur.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
Hi All, I've set up an Asterisk box with a SIP trunk to our PSTN provider. I've configured two incoming phonenumbers. One phonenumber is for voice-calls, the other one for receiving faxes. I want the incoming voice-calls to be coded by the G.729 codec, and the fax-number by G.711. Can I make a codec-negotation based on the called number? If you need more info on this, i can send it to you. Thank you all for your answer(s)! Regards, Ronald Voermans -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060209/61e3d94d/attachment.htm
Florian, What exactly do you mean by seperating traffic in to differt SIP peers? The situation is as follows: I have OpenSer connected to our SIP provider/PSTN Provider (the answer to your question: Enertel). Asterisk registers to OpenSer, which then forwards the call to PSTN. Asterisk registers two numbers at OpenSer; one phonenumber and one faxnumber. I also made two entries in sip.conf. However, the host=... Is the same for both numbers. So incoming calls are always matched to one (1) peer/entry in sip.conf. Hence the problem with negotiating the right codec (g.729 for voice, g.711 for fax). Met vriendelijke groet, ----------------------------------------------- R.L.L.M. Voermans Access & Hosting E-mail: r.voermans@global-e.nl Tel.: +31 (0)161 - 88.88.88 Fax: +31 (0)161 - 88.88.99 Global-e Raadhuisstraat 32 5126 CJ GILZE http://www.global-e.nl ----------------------------------------------- -----Oorspronkelijk bericht----- Van: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Namens Florian Overkamp Verzonden: donderdag 9 februari 2006 18:27 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Codec negotiation Hi, Ronald Voermans wrote:> I've set up an Asterisk box with a SIP trunk to our PSTN provider. > I've configured two incoming phonenumbers. One phonenumber is for > voice-calls, the other one for receiving faxes. I want the incoming > voice-calls to be coded by the G.729 codec, and the fax-number byG.711.> Can I make a codec-negotation based on the called number?Nope, but maybe you could separate the traffic in to different SIP peers.> If you need more info on this, i can send it to you.If you want we could figure something out. Just curious: Which PSTN provider are you using ? Florian _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Yes, But without going deeper into OpenSer (since this IS a Asterisk list): With OpenSer I'm using RTPPRoxy. I don't think i can manage rtpproxy to bind to multiple addresses. I'll look for that anyway. Thanks, Regards, Ronald. -----Oorspronkelijk bericht----- Van: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Namens Florian Overkamp Verzonden: donderdag 9 februari 2006 23:38 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Codec negotiation Hi Ronald, Ronald Voermans wrote:> What exactly do you mean by seperating traffic in to differt SIPpeers?> > The situation is as follows: > > I have OpenSer connected to our SIP provider/PSTN Provider (the answer> to your question: Enertel).Ah 'kay.> Asterisk registers to OpenSer, which then forwards the call to PSTN. > Asterisk registers two numbers at OpenSer; one phonenumber and one > faxnumber. I also made two entries in sip.conf. However, the host=... > Is the same for both numbers. So incoming calls are always matched to > one > (1) peer/entry in sip.conf. Hence the problem with negotiating the > right codec (g.729 for voice, g.711 for fax).Hrm, yes for inbound the problem is with the host=.. matching. Maybe Olle has a good suggestion on this :-P. However, if you control the OpenSer yourself you could easily bind another IP, or perhaps use OpenSer rules to do the trick ? Asterisk SIP stack doesn't seem suited for this type of traffic separation I guess... Florian _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
We have four settings for the codec. How will it be negotiated? How should it be negotiated in relation to the available bandwidth? Is there an influence by using canreinvite=yes ? Phone A has a setting for the priority of codec Sip.conf has (maybe even different) settings for the priority of codec of this phone A Sip.conf has codec settings for the destination phone B Phone B has (maybe even different) settings for the priority of codec Which codec will be taken? a. if the call goes via * ? b. if the call will be completed with canreinvite=yes ? Which codec should be enforce depending on the bandwidth? Thanks for thinking with me ;-) bye Ronald Wiplinger
If I have an incoming call which uses G.711u, which then gets transferred to a phone which has G.729 selected as its first preference (with 711 as a third). Is it normal behaviour for asterisk to transcode the call to G.729 rather than keep it as 711? Does anyone know if when T.38 support is complete, it can be treated as if it were a codec, ie. can put allow=g729 allow=t38 in sip/iax.conf ?
I have two polycom phones. One on a slow link, and one on a fast one. I'm trying to set the phone on the slow link to use G729 as it's first preference, and the phone on the fast link to use G711 as it's first preference. sip.conf has: [general] allow=ulaw allow=g729 [slow-link] ; Override codecs for slow link phone. allow = g729 allow = ulaw When the slow link phone dialls the fast link phone, it sends G729 as it's first preference in the INVITE to Asterisk. Asterisk then sends G729 as the first preference in the INVITE to the fast link phone. Why doesn't Asterisk send G711 instead? This raises an interesting question. If one phone uses G729, and one G711, then Asterisk is going to have to transcode, and I am going to use up a G729 license. It would seem more beneficial for it to work the way it is now. That is, both legs are using G729. Why is this better? It doesn't chew up a G729 license as there is no transcoding, and heck, if one of your call legs is G729, then the G711 party isn't going to hear anything better anyway. Thoughts?
> -----Original Message----- > From: Martin Joseph [mailto:ast@stillnewt.org] > Sent: Monday, July 17, 2006 11:40 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Codec Negotiation > > > > On Jul 17, 2006, at 9:25 AM, Douglas Garstang wrote: > > > I have two polycom phones. One on a slow link, and one on a > fast one. > > I'm trying to set the phone on the slow link to use G729 as > it's first > > preference, and the phone on the fast link to use G711 as > it's first > > preference. > > > > sip.conf has: > > [general] > > allow=ulaw > > allow=g729 > > > > [slow-link] ; Override codecs for slow link phone. > > allow = g729 > > allow = ulaw > > > > When the slow link phone dialls the fast link phone, it > sends G729 as > > it's first preference in the INVITE to Asterisk. Asterisk > then sends > > G729 as the first preference in the INVITE to the fast link > phone. Why > > doesn't Asterisk send G711 instead? > Because you set the calling to prefer g729? What did you expect?I expected Asterisk to send G711 instead, as that's what is set in [general] in sip.conf
I'm a little confused about Asterisk codec negotiation. Hopefully someone can help. I have two phones, one on a slow link where I'd like to use G729, and one on a fast link where I'd like to use ulaw. My sip.conf has: [general] allow=ulaw allow=g729 ... [slow-phone] allow=g729 allow=ulaw Firstly, does setting the codec for the slow-link phone override the general settings? Of course it's not actually documented anywhere. When the fast link phone calls the slow link phone, it sends ulaw and G729 in that order to Asterisk. When Asterisk relays the INVITE to the slow link phone, it does not change the codec preference, and sends ulaw followed by G729. I end up with a call that's ulaw on both legs, which isn't what I want. I guess the settings in [slow-phone] aren't overriding the settings in [general]. That's bad... How can I work around this? Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060720/6ef61f38/attachment.htm
Marty, Ahhh.... I wasn't thinking about the fact that it would be keyed of the callers settings, rather than the callee's. However, setting the slow-link phone to g729 isn't a very workable solution. We want to have ulaw as a backup, in case all of our g729 licenses are in use. Having the call completely fail in this case would be very bad. We should be able to have the slow-link phone negotiate to ulaw. Doug. -----Original Message----- From: Martin Joseph [mailto:ast@stillnewt.org] Sent: Thursday, July 20, 2006 11:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Codec Negotiation On Jul 20, 2006, at 10:16 AM, Douglas Garstang wrote: I'm a little confused about Asterisk codec negotiation. Hopefully someone can help. I have two phones, one on a slow link where I'd like to use G729, and one on a fast link where I'd like to use ulaw. My sip.conf has: [general] allow=ulaw allow=g729 ... [slow-phone] allow=g729 allow=ulaw Firstly, does setting the codec for the slow-link phone override the general settings? Of course it's not actually documented anywhere. When the fast link phone calls the slow link phone, it sends ulaw and G729 in that order to Asterisk. When Asterisk relays the INVITE to the slow link phone, it does not change the codec preference, and sends ulaw followed by G729. I end up with a call that's ulaw on both legs, which isn't what I want. I guess the settings in [slow-phone] aren't overriding the settings in [general]. That's bad... How can I work around this? As you already stated in your previous post the slow phone codec pref does override general when it's the caller. I think the calling parties codec preferences are respected. That is why I suggested the last time you posted this that you "force" the slow link to g729 (allow that only), as that will cause the calling party (fast) to choose g729 also... I remember reading this described somewhere, but can't find the docs at the moment. HTH, Marty -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060720/e2139fd5/attachment.htm
Sorry for the top posting. My email client is misbehaving. Can't use gsm. The polycom phones only support g711/ulaw and g729. No, we aren't intending to check for available g729 codecs.... that's why we wanted to have ulaw as a backup when no g729 codecs where available. -----Original Message----- From: Martin Joseph [mailto:ast@stillnewt.org] Sent: Thursday, July 20, 2006 12:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Codec Negotiation On Jul 20, 2006, at 11:00 AM, Douglas Garstang wrote: Subject: Re: [asterisk-users] Codec Negotiation On Jul 20, 2006, at 10:16 AM, Douglas Garstang wrote: I'm a little confused about Asterisk codec negotiation. Hopefully someone can help. I have two phones, one on a slow link where I'd like to use G729, and one on a fast link where I'd like to use ulaw. My sip.conf has: [general] allow=ulaw allow=g729 ... [slow-phone] allow=g729 allow=ulaw Firstly, does setting the codec for the slow-link phone override the general settings? Of course it's not actually documented anywhere. When the fast link phone calls the slow link phone, it sends ulaw and G729 in that order to Asterisk. When Asterisk relays the INVITE to the slow link phone, it does not change the codec preference, and sends ulaw followed by G729. I end up with a call that's ulaw on both legs, which isn't what I want. I guess the settings in [slow-phone] aren't overriding the settings in [general]. That's bad... How can I work around this? As you already stated in your previous post the slow phone codec pref does override general when it's the caller. I think the calling parties codec preferences are respected. That is why I suggested the last time you posted this that you "force" the slow link to g729 (allow that only), as that will cause the calling party (fast) to choose g729 also... I remember reading this described somewhere, but can't find the docs at the moment. HTH, Marty Marty, Ahhh.... I wasn't thinking about the fact that it would be keyed of the callers settings, rather than the callee's. However, setting the slow-link phone to g729 isn't a very workable solution. We want to have ulaw as a backup, in case all of our g729 licenses are in use. Having the call completely fail in this case would be very bad. We should be able to have the slow-link phone negotiate to ulaw. Are you intending to implement some logic to check for available g729 codecs? Because asterisk doesn't do this for you... What about using some form of unrestricted codec like GSM instead for the slow link? Don't know any great solutions for you... Marty -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060720/52f42e81/attachment.htm
Can't put it in a realtime database. We have multiple Asterisk boxes in a cluster, and it's a well known fact that multiple Asterisk boxes using realime cannot query a common MySQL database. Sounds crazy, but true. Doug.> -----Original Message----- > From: Marco Mouta [mailto:marco.mouta@gmail.com] > Sent: Friday, July 21, 2006 4:14 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Codec Negotiation > > > Just an idea: > > Put this Slow-Phone sip account into sip realtime database, and > outside of asterisk manage to verify G729 licenses availability and > script it to your SIP-realtime. > > This way every call to this SIP account will go to SIP realtime > database that is being changed by an external script that just > monitors your G729 licences, and keeps on track which codec is going > to be used: Ulaw or G729. > > Don't know if this is a good idea, just a suggestion. > > Best regards, > Marco Mouta > > On 7/21/06, Woodoo People .pGa! <wpeople@shadow.microsystem.hu> wrote: > > > >No, we aren't intending to check for available g729 codecs.... > > > >that's why we wanted to have ulaw as a backup when no g729 codecs > > > >where available. > > > > > > > That won't work. If it's trying to use G729, it will > still try even > > > when the licenses are all in use. So you need to either > force it g729 > > > and make sure there are always licenses for it available, > or use ulaw > > > and make sure there is enough bandwidth. > > > > > > The other option is to write your own code that checks to > verify the > > > licenses are free somehow, and then tampers with the codec > > > preferences? I think Brett (trixter) has some ideas/work in this > > > direction already. > > > > i heard somewhere, when g729 licences are gone, it will > work as g711, > > is this info FAKE? > > > > > > -- > > WoodOO-[P]an[G]alaktikan[A]gent-People <][> http://shadow.pganet.com > > > wpeople@shadow.pganet.com]iCQ#33118021[wpeople.on.iRCNet]wpeop > le@RedHat.users > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > Com os melhores cumprimentos, > > Marco Mouta > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
> -----Original Message----- > From: Brian Capouch [mailto:brianc@palaver.net] > Sent: Friday, July 21, 2006 11:33 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Codec Negotiation > > > Douglas Garstang wrote: > > Can't put it in a realtime database. We have multiple > Asterisk boxes in a cluster, and it's a well known fact that > multiple Asterisk boxes using realime cannot query a common > MySQL database. Sounds crazy, but true. > > > > You spread some amazing "well-known facts" on this list. As usual > salted with your typical choice of words that implies that > Asterisk has > "crazy" flaws that no sane programmer would countenance. > > I have a dozen or more Asterisk boxes that all query the exact same > Realtime database. The setup works fine, and the "time to > deploy" a new > station with very elaborate functionality is reduced to minutes. The > ability to rearrange behaviors on the fly is also a great feature. I > love ARA. > > I use Postgres and not MySQL, but I can't believe that the > choice is SQL > engine would make a difference. > > I think you confuse the requirements of your deployment > scenario, which > a few minutes ago on this list you yourself characterized as > "ridiculous," with underlying common features of Asterisk used in > quotidian circumstances.Would you like me to dig up the posts from Keving Fleming stating that this is known not to work Brian?
> -----Original Message----- > From: Brian Capouch [mailto:brianc@palaver.net] > Sent: Friday, July 21, 2006 12:08 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Codec Negotiation > > > Douglas Garstang wrote: > > > > > > > Would you like me to dig up the posts from Keving Fleming > stating that this is known not to work Brian? > > As I recall those posts have to do with the way your particular setup > required ARA to work with a failover/redundant cluster system > you were > building. > > Beyond that I'm not really interested in getting into a > pissing contest. > I have ONE SQL table called "extensions_table" on ONE SQL > server, but > have maybe 20 SIP phones using that same database, placing calls from > 10-12 separate Asterisk instances. > > I was calling into question your presenting a "well-known fact" that > appears to be incorrect. If Kevin sees this and wants to chime in to > support your statement and tell me that my experience is somehow an > illusion, he's certainly welcome to do so. I have > experienced the taste > of crow, and eat it when needed. You? > > Can certain situations be construed where ARA will not do > exactly what > the administrator wants? Apparently, from reading some of > your posts, true. > > Can multiple Asterisk servers be set up to use a single database > instance to store common configuration information? Certainly true, > from my and many other people's experiences. > > The thrust of my post was to refute the "fact," and to suggest you > perhaps adopt a little less inflammatory rhetoric when you > post to this > list.Here's a post Kevin Fleming made to the group on Dec 12, 2005, in response to my question on this subject. I was quite clear about my question, and does not involve clustering, only N number of Asterisk boxes all pointed to the same database. It seems his reply is also quite clear. It's good that it's working for you. I guess you got lucky. "Douglas Garstang wrote:> Thanks.... while we're on the topic of realtime. Can realtime sipusers be shared amongst multiple Asterisk boxes, to share a common location database? I'm sitting here on a Sunday jerking around with it, having problems. I'd like to know before I spend more Sundays doing the same thing if it's even supposed to work or not.Uhhh... you already quoted my previous message on that topic stating that it was not supported at this time. In any given situation, it may or may not work properly, depending on exactly what the servers and clients are doing. Even if the code had been written, there will still be many issues involved in actually implementing it, including (but not limited to) NAT traversal, call limit handling, registration expiration and others. It also mandates that there can be _no_ caching of peer/user information in memory, which currently means there is no 'qualify' or MWI notification possible." Doug
Well, I wish someone would tell Kevin Fleming that.> -----Original Message----- > From: olivier.taylor [mailto:olivier.taylor@gmail.com] > Sent: Friday, July 21, 2006 1:04 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Codec Negotiation > > > I must agree, > > we use 2 Ser in front of 4 asterisk sharing the same database cluster. > > Olivier > > Brian Capouch a ?crit : > > Douglas Garstang wrote: > > > >> > >> > >> Would you like me to dig up the posts from Keving Fleming stating > >> that this is known not to work Brian? > > > > As I recall those posts have to do with the way your > particular setup > > required ARA to work with a failover/redundant cluster > system you were > > building. > > > > Beyond that I'm not really interested in getting into a pissing > > contest. I have ONE SQL table called "extensions_table" on ONE SQL > > server, but have maybe 20 SIP phones using that same > database, placing > > calls from 10-12 separate Asterisk instances. > > > > I was calling into question your presenting a "well-known > fact" that > > appears to be incorrect. If Kevin sees this and wants to > chime in to > > support your statement and tell me that my experience is somehow an > > illusion, he's certainly welcome to do so. I have experienced the > > taste of crow, and eat it when needed. You? > > > > Can certain situations be construed where ARA will not do > exactly what > > the administrator wants? Apparently, from reading some of > your posts, > > true. > > > > Can multiple Asterisk servers be set up to use a single database > > instance to store common configuration information? > Certainly true, > > from my and many other people's experiences. > > > > The thrust of my post was to refute the "fact," and to suggest you > > perhaps adopt a little less inflammatory rhetoric when you post to > > this list. > > > > B. > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
> -----Original Message----- > From: Joshua Colp [mailto:jcolp@digium.com] > Sent: Friday, July 21, 2006 9:38 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Codec Negotiation > > > ----- Original Message ----- > From: Douglas Garstang > [mailto:dgarstang@oneeighty.com] > To: olivier.taylor@phonext.com, Asterisk > Users Mailing List - Non-Commercial Discussion > [mailto:asterisk-users@lists.digium.com] > Sent: Fri, 21 Jul 2006 16:21:15 > -0300 > Subject: RE: [asterisk-users] Codec Negotiation > > > > Well, I wish someone would tell Kevin Fleming that. > > > > So... the realtime architecture can work when shared between > servers - but not all the time, and it depends on certain > variables. Multiple servers can grab the same information > from the database, but it doesn't mean they will be able to > contact the phone for example. This is what Kevin was talking > about. If a phone is behind NAT and registers to one Asterisk > machine, the device doing the NAT will setup in it's memory a > mapping of the external port to the internal IP+port. > Depending on the implementation and setup, only the server > that the phone registered to will be able to send packets > back. Thus if you have failover to another server for > example, that server will not be able to contact the phone. > > Like I said, there's too many variables to make it work all > the time... > > Now - you can pull information in order for them to be able > to place calls. That should be fine.Don't know about that. I'm not sure why he brought NAT up, as I didn't mention it. Might be a red herring I think.