search for: chan_pjsip

Dne 13.8.2015 v 17:20 Rusty Newton napsal(a): > On Thu, Aug 13, 2015 at 3:54 AM, Marek ?ervenka <cervajs at fpf.slu.cz > <mailto:cervajs at fpf.slu.cz>> wrote: > > hello, > > is it possible simultaneously use chan_sip and chan_pjsip? > > if yes, can you recommend settings > > i'm thinking about > - chan_sip - for sip hardphones/softphones (sip udp 5060) > - chan_pjsip - for webrtc > > > You can use both.. you will want to make sure your bind addresses and > ports don't co...

hello, is it possible simultaneously use chan_sip and chan_pjsip? if yes, can you recommend settings i'm thinking about - chan_sip - for sip hardphones/softphones (sip udp 5060) - chan_pjsip - for webrtc -- --------------------------------------- Marek Cervenka =======================================

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...

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) -...

...G[23961][C-00000000]: translate.c:490 ast_translator_build_path: No translator path: (ending codec is not valid) [Oct  2 07:24:55] WARNING[23961][C-00000000]: translate.c:490 ast_translator_build_path: No translator path: (starting codec is not valid) [Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw frame when native formats are (siren7) (rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000)) [Oct  2 07:24:55] WARNING[2396...

Hi, We have been migrating our PBX system from Asterisk 1.8 and chan_sip to Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have stumbled on a behaviour difference I don't like. With chan_pjsip when a phone went unexpectedly offline (Ethernet cable disconnected) Asterisk would detect this quickly (through the 'qualify' pings), mark the phone as 'Unavailable' and fai...

...14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module 'func_pjsip_contact.so': /usr/lib64/asterisk/modules/func_pjsip_contact.so: undefined symbol: ast_sip_get_contact_status_label [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module 'chan_pjsip.so': /usr/lib64/asterisk/modules/chan_pjsip.so: undefined symbol: ast_sip_cli_traverse_objects and so on. module show like res_pjproject Module Description Use Count Status Support Level res_pjproject.so PJPROJECT Log and Utility Support 1...

...is not high enough. I need a solution, too. I understand that this behavior is a nogo if you have a lot of calls because transcoding is expensive. Thanks Michael On 05.07.23 at 17:58 Michael Ulitskiy wrote: > 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 t...

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 phreak...

...mething. What are your thoughts? Thanks, *Michael Ulitskiy* Ace Innovative Networks, Inc. Main/SMS: 212-868-2366 Direct/SMS: 212-812-1203 https://www.aceinnovative.com On 7/5/23 11:58, Michael Ulitskiy wrote: > > 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 tha...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Le 21/05/2015 00:16, Joshua Colp a ?crit : > If CCSS is needed then the only option is to use chan_sip. The > chan_pjsip module does not implement CCSS in any way. Is CCSS support planned for PJSIP? chan_sip is in "extended" state in asterisk-13, so chan_pjsip should be preferred for new installations, ri ght? Thanks, - -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran...

Hey all! I recently tried the dtmf_mode "auto_info" on my setup to support endpoints that only understand SIP INFO as a fallback. My setup is the following: Endpoint A (RFC4733) --> Asterisk <-- Endpoint B (SIP INFO) Both are configured with "auto_info" dtmf_mode in pjsip.conf. What I ran into is, that DTMF sent from endpoint A to endpoint B is additionally sent via

...E to Unreachable devices'. Jacek > On Tue, Aug 9, 2016 at 12:03 PM, Jacek Konieczny <jajcus at jajcus.net > <mailto:jajcus at jajcus.net>> wrote: > > Hi, > > We have been migrating our PBX system from Asterisk 1.8 and chan_sip to > Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have > stumbled on a behaviour difference I don't like. > > With chan_pjsip when a phone went unexpectedly offline (Ethernet cable > disconnected) Asterisk would detect this quickly (through the 'qualify' > pings), mark th...

Hello. I am using asterisk and chan_sip a lot of years. And newbie in chan_pjsip. Now i am transfering all from chan_sip to chan_pjsip. And have a lot of questions. First of... system: Asterisk 13.2 on slackware 14.1 Errors on outgoing call: [2015-03-03 00:18:58] ERROR[6794]: chan_pjsip.c:1778 request: Failed to create outgoing session to endpoint 'srv_d228' [2015-03...

...irard <jd.girard at sysnux.pf > <mailto:jd.girard at sysnux.pf>>: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Le 21/05/2015 00:16, Joshua Colp a ?crit : > > If CCSS is needed then the only option is to use chan_sip. The > > chan_pjsip module does not implement CCSS in any way. > > Is CCSS support planned for PJSIP? chan_sip is in "extended" state in > asterisk-13, so chan_pjsip should be preferred for new installations, ri > ght? > > > If you really want CCSS support and to be fancy wit...

Ok so now I'm getting this when doing a make in asterisk... travis at pcimphone1:~/downloads/asterisk-13.5.0$ make [LD] chan_pjsip.o pjsip/dialplan_functions.o -> chan_pjsip.so /usr/bin/ld: /usr/local/lib/libpjsip-ua-x86_64-unknown-linux-gnu.a(sip_inv.o): relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC /usr/local/lib/libpjsip-ua-x86_64-unknown-linux-gnu.a: erro...

...> Unable to load module res_pjsip > Command 'module load res_pjsip' failed. > ERROR[141535]: loader.c:281 module_load_error: Error loading module > 'res_pjsip': /usr/lib64/asterisk/modules/res_pjsip.so: undefined symbol: > ast_statsd_log_full_va > > module load chan_pjsip > Unable to load module chan_pjsip > Command 'module load chan_pjsip' failed. > ERROR[141780]: loader.c:281 module_load_error: Error loading module > 'chan_pjsip': /usr/lib64/asterisk/modules/chan_pjsip.so: undefined > symbol: ast_sip_cli_traverse_objects > > /u...

On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy <mulitskiy at acedsl.com> wrote: > 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

Oh, that's great. It wasn't clear from that page, at least not for me. :-( Having it clearly stated on the document would save me (and probably others) lots of time. Thanks for clarifying it. Any idea on the timeframe of implementation? *Michael Ulitskiy* Ace Innovative Networks, Inc. Main/SMS: 212-868-2366 Direct/SMS: 212-812-1203 https://www.aceinnovative.com On 7/6/23 12:47, Joshua

Hi Everyone, We're routing calls through Asterisk (dialing in via sip and then dialing out via SIP). We've noticed a curious behavior in chan_sip that doesn't persist with chan_pjsip. When examining the packet capture, we're seeing the SSRC changing constantly on the call. At first it happens over a variable interval (15s 6s etc) but eventually it ends up changing exactly every 1000ms. Every time the SSRC changes, it causes a very minor but noticeable gap in audio. The f...