search for: ulaw

Let's say I have two devices configured and the follow call scenarios occur. [100] disallow=all allow=g722&ulaw Polycom phone with g722,ulaw,alaw,g729 [101] disallow=all allow=ulaw Polycom phone with g722,ulaw,alaw,g729 101 dials 100 -> ulaw to ulaw is chosen 100 dials 101 -> g722 to ulaw is chosen Ideally when 100 dials 101 ulaw would be chosen since it is the common format. Looking into this dee...

...the Asterisk documentation/book and haven't been able to find an answer. My question is: Can I get my IP phone to select a different codec depending on the final destination of each call? I've got these things connected to my Asterisk box: - Snom 300 phone (supports g729 and ulaw) - PSTN Peer (supports g729 and ulaw) - Remote Asterisk Peer (supports speex and ulaw) Currently, it's configured like this: [snom300] disallow=all allow=ulaw [pstnpeer] disallow=all allow=ulaw [asteriskpeer] disallow=all allow=speex which translates to th...

...Anyone? I have hard time to believe this is not possible with chan_pjsip. Anyway, may I ask how people handle the following scenario which I imagine should be quite common: - I have internal extensions talk to each other using g722. so their codec setting (with chan_sip now) is "allow=g722,ulaw" - I have carriers trunks that handle ulaw only (allow=ulaw) - calls between internal extensions naturally happen over g722 as its their preferred codec - for external calls I now set SIP_CODEC_INBOUND=ulaw to influence codec selection on calling channel and the calls set up using ulaw end-t...

Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7...

...     : prefer:pending, operation:intersect, keep:all, transcode:allow  codec_prefs_outgoing_offer         : prefer:pending, operation:union, keep:all, transcode:allow All endpoints have the same default config above. Let's go over simplest scenario: A calls B. A is configured with g722 and ulaw (allow=!all,g722,ulaw) and B is configured with ulaw and alaw (allow=!all,ulaw,alaw) 1. codec_prefs_incoming_offer: A sends INVITE to asterisk with codecs in SDP g722,g729,g711u,g711a: ... m=audio 2266 RTP/AVP 9 18 0 8 101. a=rtpmap:9 G722/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpm...

> On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: >> I receive an INVITE/SDP containing: >> >> m=audio 11310 RTP/AVP 3 0 101 >> >> which I interpret as gsm, ulaw, rfc2833. >> >> and I reply with an OK/SDP containing: >> >> m=audio 15884 RTP/AVP 0 3 101 >> >> which I interpret as ulaw, gsm, rfc2833. >> >> How can I tell which codec was actually used for the call? On Fri, 11 May 2018, Daniel Tryba wrote: >...

Hi! As the upstream of my DSL-connection is very slow, I'd like my sip-phones to use iLBC to connect to my *. My gateway provider only allows ulaw. Hence, I'd like to use the follwing setup: SIP-phone <--iLBC--> Asterisk <---ulaw----> PSTN-Gateway I get the following error: "Unable to find a codec translation path from ilbc to ulaw" Setup SIP-phone: disallow=all allow=ilbc Setup PSTN-Gateway: disallow=all allow=...

We are getting the following on 1.8.3 and 1.8.4-rc2, HELP! Why is Asterisk unable to transcode to/from ulaw and g722? [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while nat...

Hi list! When I receive an incoming call from a SIP peer where I've configured disallow=all allow=alaw (and no other codec) I can see the following NOTICE on the console: Dropping incompatible voice frame SIP/peer07-0000007c of format ulaw since our native format has changed to (alaw) My question is: where can I change the native format from ulaw to alaw (or something else)? Is ulaw, as the native format, adjustable in some config file or is it hard-coded into Asterisk? It doesn't seem to have any effect on the voice quality...

Hi List, The below error caused the Asterisk to crash, if anyone have idea on this please reply,(Asterisk version :1.8.9) [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to 0x400 (ilbc) [Sep 2 15:59:53] WARNING[24418] chan_sip.c: Asked to transmit frame type ilbc, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) [Sep 2...

Ok, My provider is sending a call to me via ULaw but Asterisk isn't picking up on this, I've only allowed ulaw, I disallow=all and then allow=ulaw in my sip.conf and that's the only thing I allow, but when my provider sends me the requests, I get an error about No Compatible Codecs: 17 headers, 8 lines Using latest request as...

...ber without and. -- Executing [888 at from-internal:1] Set("SIP/103-0000035d", "LANGUAGE=en") in new stack -- Executing [888 at from-internal:2] SayNumber("SIP/103-0000035d", "1234") in new stack -- <SIP/103-0000035d> Playing 'digits/1.ulaw' (language 'en') -- <SIP/103-0000035d> Playing 'digits/thousand.ulaw' (language 'en') -- <SIP/103-0000035d> Playing 'digits/2.ulaw' (language 'en') -- <SIP/103-0000035d> Playing 'digits/hundred.ulaw' (language 'en...

I'd like to use native moh instead of with mpg123... for some reason the processes never bloody die. For native moh to not spawn an external player, I'd need to convert the default supplied moh sound files in /var/lib/asterisk/mohmp3 to ulaw and g729 format. Anyone know of a free, easy way to convert them? Thanks, Doug.

Hi everyone, I'm having an odd issue. I've been doing some testing over the past couple weeks on some Asterisk modules / utilities, but have bumped into a problem which I can't seem to resolve. Asterisk can't seem to play the default sound files (ULAW) in my environment. All necessary debugging information is included below. I'd love to get anyone else's thoughts on this, what I may be doing wrong, or how to resolve the issue. Thanks so much! Asterisk Version: Asterisk 1.6.1.11 OS: ubuntu-sever 10.04, 64-bit Kernel: 2.6.32-24-server #39...

Hi all, i have a little problem to understand this warning message, it's annoying and it cause a lot of spurious in the log files. Im working with this scenario: a sip outgoing trunk (Trunk-out) that support only ulaw and all calls are always routed to this. a list of sip UAs that potentially can use any codec apart g729/g723. I setup the asterisk to do as mediaproxy so directmedia=no and canreinvite=no. I want the rtp traffic goes through asterisk. I can reproduce the waring message below when a peer uses a di...

Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw => ulaw is choppy, ulaw => alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit-

...ssthrough coders as ATA is capable of both coders. (I think even if you have the licenses, * should try avoid codec-conversions when ever it can) Here is my settings in sip.conf. I will only list the required codec related lines, for easy understanding, [general] disallow=all allow=g729 allow=ulaw allow=alaw register => sip-a@foo.com register => sip-b@bar.com [sip-a] .... disallow=all allow=ulaw [sip-b] ... disallow=all allow=g729 [ATA] ..... canreinvite=no Here is what happens when I look at the SIP packets from linux. (ethereal) Case 1 : ATA Dialing out through sip-a ========...

I have many endpoints and each endpoint has some parameter in common so i wonder is there any way to config one for all endpoints? Like in my example I have two endpoints and I repeat the same thing, [100] type=endpoint aors=100 auth=100-auth allow=ulaw,alaw,gsm,g726 context=from-internal callerid=device <100> dtmf_mode=rfc4733 use_avpf=no ice_support=no media_use_received_transport=no trust_id_inbound=yes send_pai=yes rtp_symmetric=yes rewrite_contact=yes message_context=astsms [200] type=endpoint aors=200 auth=200-auth all...

...terisk in Sydney. The call is to asterisk in New York. The caller in NZ can hear clearly. Nothing in NY. Here's the sydney server: -- Accepting AUTHENTICATED call from <zoiperipaddr>: > requested format = speex, > requested prefs = (), > actual format = ulaw, > host prefs = (silk16|ulaw|gsm|g722), > priority = mine -- Executing [8447 at nz-in:1] Dial("IAX2/n4-270", "IAX2/sydney") in new stack -- Called IAX2/sydney -- Call accepted by <nyipaddr> (format ulaw) -- Format for call is (ula...

I'm trying to force GSM when I call on certain extension but I'm getting connected with "ulaw" Which is not suitable when bandwidth is low and slow. my phone is iax-322 in iax.conf [iaxy-322] ... disallow=all allow=gsm allow=ulaw allow=alaw [zoiper_kathy_old_phone] ... disallow=all allow=gsm allow=ilbc allow=ulaw allow=alaw allow=speex I've define "allow=gsm" in as...