similar to: Why doesn't Asterisk try to prevent transcoding

Hello, 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

Hello Michael, you are referring to the following behavior - did I get it correctly?: outbound broken: asterisk offers g722 / g711 to provider (callee), callee answers g711. Asterisk now transcodes between caller and callee (g722 <-> g711). inbound works: call from provider: g711 -> asterisk drops g722 and passes g711 to internal callee -> no transcoding. As far as I know,

Hello, I finally got to look at chan_sip to chan_pjsip migration again. This time I’m having problems with influencing codec selection on originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only works on outbound (called) channel and has no affect on calling channel. My experiments and function documentation (which says “Media and codec offerings to be set on an outbound SIP

We want to record inbound channels at 16kHz, but send only 8kHz to our peers. I've set our default profile in sip.conf to disallow all but g722, and the peers disallow all but ulaw. We have a proxy in front of Asterisk that is configured to disallow all but G722 also. My test calls show inbound to the proxy is recorded at 16kHz, inbound in Asterisk is only 8kHz, and the peers receive 8kHz. So

Hello, I'd like to ask whether there is some documentation with recommended parameters for transcoding voice codecs such as G722, G711a/u <-> Opus with near-transparency. My Idea is to have something like: HW-Phone <-> Asterisk <---------> Asterisk <-> HW-Phone (G722) (Opus) (G722) in order to lower the bandwidth between the two

Hello guys, i have a question about a voip gateway we use. I saw those values typing in cli: core show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - - - - - - - - - - - - gsm - - 2001 2001 6000 2001 2000 16000 - 34002 - 6000

I am unable to get re-invite to work on a new system. Also, unwanted transcoding is occurring on PSTN calls. The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2, CentOS 5.8) currently in production. Both systems are on VPS with public IP addresses. Goals for the new system include: HD (g722) connections on

Well, I'm trying to migrate to chan_pjsip so that I don't have to do that. It's so surprising that the issue so seemingly obvious and trivial hasn't been addressed yet that I wanted to query the collective wisdom of this list to verify my observations. Thanks for github pointer. Michael On 7/5/23 16:46, asterisk at phreaknet.org wrote: > On 7/5/2023 4:19 PM, Michael

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,

What is the current best practice to avoid transcoding on an outgoing call to a party whose codec preference is not known in advance? In other words, incoming calls are easy since codecs are negotiated from least-known (the remote party) to most-known (my endpoint) and my codecs can simply be preferred accordingly to match the remote. Outbound calls seem harder. Our endpoints always negotiate

Today my pbx stopped responding to my sip phones.. looking into the log, here what I got: Jun 23 15:50:05 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new): Unable to allocate socket: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 1308 (sip_alloc): Unable to create RTP session: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 4655

Hello, After I have re-read the "PJSIP Advanced Codec negotiation" document, it occurred to me that the desired behavior should actually happen automatically, just due to the codec negotiation logic, but it looks like asterisk doesn't actually follow the described logic which is likely a bug. Can you please follow with me through a simple sip call and see if I'm missing

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000

Hello! I'm facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package). Background: The call is initiated by asterisk and is received by the same asterisk conference room via Phone extension -> asterisk -> provider A -> provider B -> asterisk. Asterisk initially sends

Hello, we are curently benchmarking an asterisk system 1034 sip users are logged into this system and the test software is trying to establish 400 concurrent calls. In the CLI I see the following messages: Apr 19 14:20:51 WARNING[4045]: rtp.c:911 ast_rtcp_new: Unable to allocate socket: Too many open files Apr 19 14:20:51 WARNING[4045]: acl.c:306 ast_ouraddrfor: Cannot create socket Apr 19

Hello, I have some problems to use G722, when my client sent an invite request to asterisk using G722/16000 codec asterisk respond with G722/8000 codec. I dont know exactly if Asterisk supports G722/16000 codec?? If yes how can I activate It?? Thanks. Rachid. Below wireshak trace:

Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723

Hi! I want to tell asterisk to simply pass-through any codecs that my phones support. I have to use codecs that are not popular and implemented by a third-party, asterisk has nothing to do with them. I've made a test with g722 (that asterisk doesn't support), i've set all my two snom 300 phones to support only g722 and asterisk declined the sip invitation. That is bad for me. Is it

I have Polycom phones that support the g722 codec. Adding allow=g722 to the [general] section of sip.conf works great and I can make calls between the phones using g722. However Asterisk is negotiating g722 for calls going out my voip provider and transcoding these to ulaw. In sip.conf for the provider I have deny=all and allow=ulaw. This can cause potential audio degrading and wastes cpu cycles.

Hello, I am trying to enable call forwarding feature on asterisk 1.6.0.9 with asterisk-gui. Sure there is no menu for that on gui but, when i try to write some example scripts to extensions.conf to make it work. I totally failed. I dont wanna install smthing like freepbx etc on the system so, i need help to add unconditional etc call forwarding feature for 1.6 Thanks