search for: ast_channel_nam

...ot;ZAP" channel ... > > and the code says > > ast_log(LOG_WARNING, "No %s channel available for conference, user > introduction disabled\n", dahdi_chan_name); > > and > > in main/asterisk.c > > > #ifdef HAVE_ZAPTEL > static char _dahdi_chan_name[AST_CHANNEL_NAME] = "Zap"; > #else > static char _dahdi_chan_name[AST_CHANNEL_NAME] = "DAHDI"; > #endif > > I deduct from that ... that you're still running zaptel and not dahdi. > Because your log should say "No DAHDI channel available ..." UNLESS > for some...

Hi, I am using asterisk 1.8.22 and have a problem when calling in parallel several SIP endpoints and I am not sure how to resolve it. In this case Asterisk will not bridge any audio to the caller before the 200 OK. Which means any progress announcements, including remotely generated ringback, are not passed back to the caller. This behavior is completely correct, because there is no way to know

We recently updated our Asterisk (1.4.24) box from Zaptel (1.4.12.1) to Dahdi (2.1.0.4). Everything seemed to go smooth with the exception of meetme. Meetme seems to not be able to find a zap channel for conferencing. We use voice introductions in our conference bridge and it seems to break that feature. The error from the console is.... # app_meetme.c:2593 find_conf: No Zap channel available for

Hello everyone, I am using the Asterisk REST API in order to establish a call to an endpoint and to send over a remote file (HTTP). The issue is that I am experiencing an audio quality issue. I have tried encoding the file differently, but everytime Asterisk is cutting the audio frequencies above 4Khz. The call is established with G.722 and the audio file is mono 16Khz 16 bit sln16 extension.

...832 - [patch]DTLS-crashes within openssl (Reported by Stefan Engstr??m) * ASTERISK-25127 - DTLS crashes following "Unable to cancel schedule ID" in dtls_srtp_check_pending (Reported by Dade Brandon) * ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at channel_internal_api.c (Reported by Carl Fortin) * ASTERISK-25115 - Crash related to func sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c (Reported by John Bigelow) * ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early replaces call pickup (Rep...

...ted by Jens T.) * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) * ASTERISK-26879 - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) * ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive (Reported by Ross Beer) * ASTERISK-27225 - Crash when freeing dtls_cfg->cafile (Reported by Richard Kenner) * ASTERISK-27177 - ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c...

...van de Sande) * ASTERISK-26879 - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) * ASTERISK-27165 - CDR: CDR(start,u) function won't work in cdr_custom config (Reported by Jacek Konieczny) * ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive (Reported by Ross Beer) * ASTERISK-27225 - Crash when freeing dtls_cfg->cafile (Reported by Richard Kenner) * ASTERISK-27177 - ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c...

...ted by Jens T.) * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) * ASTERISK-26879 - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) * ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive (Reported by Ross Beer) * ASTERISK-27225 - Crash when freeing dtls_cfg->cafile (Reported by Richard Kenner) * ASTERISK-27177 - ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c...

...ted by Jens T.) * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) * ASTERISK-26879 - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) * ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive (Reported by Ross Beer) * ASTERISK-27225 - Crash when freeing dtls_cfg->cafile (Reported by Richard Kenner) * ASTERISK-27177 - ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c...

...ted by Jens T.) * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) * ASTERISK-26879 - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) * ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive (Reported by Ross Beer) * ASTERISK-27225 - Crash when freeing dtls_cfg->cafile (Reported by Richard Kenner) * ASTERISK-27177 - ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c...

...ted by Jens T.) * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) * ASTERISK-26879 - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) * ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive (Reported by Ross Beer) * ASTERISK-27225 - Crash when freeing dtls_cfg->cafile (Reported by Richard Kenner) * ASTERISK-27177 - ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c...

...832 - [patch]DTLS-crashes within openssl (Reported by Stefan Engstr??m) * ASTERISK-25127 - DTLS crashes following "Unable to cancel schedule ID" in dtls_srtp_check_pending (Reported by Dade Brandon) * ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at channel_internal_api.c (Reported by Carl Fortin) * ASTERISK-25115 - Crash related to func sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c (Reported by John Bigelow) * ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early replaces call pickup (Rep...

...832 - [patch]DTLS-crashes within openssl (Reported by Stefan Engstr??m) * ASTERISK-25127 - DTLS crashes following "Unable to cancel schedule ID" in dtls_srtp_check_pending (Reported by Dade Brandon) * ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at channel_internal_api.c (Reported by Carl Fortin) * ASTERISK-25076 - res_pjsip: Failover does not occur on connection-less transport or 503 response (Reported by Joshua Colp) * ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early replaces call pickup (Reported...

...Sande) - [ASTERISK-26879 <https://issues.asterisk.org/jira/browse/ASTERISK-26879>] - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) - [ASTERISK-27236 <https://issues.asterisk.org/jira/browse/ASTERISK-27236>] - Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive (Reported by Ross Beer) - [ASTERISK-27225 <https://issues.asterisk.org/jira/browse/ASTERISK-27225>] - Crash when freeing dtls_cfg->cafile (Reported by Richard Kenner) - [ASTERISK-27177 <https://issues.aster...