search for: sip_call

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not showing speex translation on "core show translation recalc 10"....

...xact same steps on one of the branch machines, then went into sip.conf on both machines and set the codec between the branches to speex and restarted asterisk on both machines. When I try to call between the branches I get the following message: [Oct 14 12:26:09] WARNING[23308]: chan_sip.c:3024 sip_call: No audio format found to offer. Cancelling call to 42 I remember seeing a post somwhere along the way stating that the new version of speex requires a change to the Asterisk code to link against a new library or something, but I couldn't find the post again. Is there something I'm mi...

Hi all, When trying to send a FAX I got the following error: Executing [003228949469 at micho:1] Dial("SIP/028949469-08466918", "SIP/ 003228949469 at 80.169.210.181|60") in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949469 -- Couldn't call 0032234534534 at 1.1.1.1.1 Where I should define the codec other than the extension in order to succeed the call? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http:...

...iend host=XXX.XX.56.4 context=default disallow=all allow=g723 allow=g729 However, the Dial application gives the following error: -- AGI Script Executing Application: (DIAL) Options: (SIP/t564/1XXXXXX4332,,HR) == Using SIP RTP CoS mark 5 [Jun 29 13:37:37] WARNING[16693]: chan_sip.c:5518 sip_call: No audio format found to offer. Cancelling call to 1XXXXXX4332 -- Couldn't call t564/1XXXXXX332 == Everyone is busy/congested at this time (0:0/0/0) I've checked to ensure that both formats are loaded into Asterisk: voip2*CLI> module show like 729 Module...

...-- Executing Dial("SIP/4000-98ec", "SIP/4001|30|Ttm") in new stack Jul 8 20:53:57 DEBUG[409616]: app_dial.c:420 dial_exec: SIMPLE DIAL (NO URL) Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:835 create_addr: Setting NAT on RTP to 0 Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1040 sip_call: Outgoing Call for 4001 Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1139 find_user: 4001 is not a local user -- Called 4001 Jul 8 20:53:57 DEBUG[409616]: channel.c:1402 ast_prod: Prodding channel 'SIP/4000-98ec' Call from the Cisco (not working) Jul 8 20:54:50 DEBUG[229391]: chan_sip....

...Discussion Subject: [asterisk-users] FAX Hi all, When trying to send a FAX I got the following error: Executing [003228949469 at micho:1] Dial("SIP/028949469-08466918", "SIP/003228949469 at 80.169.210.181|60") in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949469 -- Couldn't call 0032234534534 at 1.1.1.1.1 Where I should define the codec other than the extension in order to succeed the call? Regards _______________________________________________ -- Bandwidth and Colocation Provide...

...ne 3428 (zt_read): DTMF digit: 1 on Zap/19-1 Oct 6 10:59:56 DEBUG[28688]: File chan_zap.c, Line 1056 (zt_enable_ec): No echocancellation requested Oct 6 10:59:56 DEBUG[28688]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 6 10:59:56 DEBUG[28688]: File chan_sip.c, Line 857 (sip_call): Outgoing Call for mvickers Oct 6 10:59:56 DEBUG[5126]: File chan_sip.c, Line 568 (__sip_semi_ack): (Provisional) Stopping retransmission (but retaining packet) on '5ecb985524e3ec232cf3e54d59674900@172.20.1.67' Request 102: Found Oct 6 10:59:57 DEBUG[5126]: File chan_sip.c, Line 568 (__s...

...registrar. Anybody know if I'm out of luck? I already looked into writing in support for adding the ability for asterisk to send 302 redirects to certain hosts when they are dialed via Dial(SIP/1231231234@pstn) and couldn't find where the initial sip_request was by the time dial executed sip_call so that I could create the proper 302 response... my C-fu is not so strong... :-)

...as i answered it dissconnects!!!! This is what i get from the Asterisk console: -- Executing Dial("OH323/R27469", "SIP/xlite1|10") in new stack Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265 create_addr: Setting NAT on RTP to 0 Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500 sip_call: Outgoing Call for xlite1 Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1633 update_user_counter: Call from user 'xlite1' is 1 out of 0 -- Called xlite1 Aug 13 10:19:03 DEBUG[245774]: chan_sip.c:840 __sip_semi_ack: -- SIP/xlite1-89a7 is ringing Aug 13 10:19:05 DEBUG[245774]: chan_sip.c:799...

...in new stack -- Executing Dial("Zap/1-1", "SIP/7011|30|r") in new stack May 27 18:08:06 DEBUG[1224]: app_dial.c:497 dial_exec: SIMPLE DIAL (NO URL) May 27 18:08:06 DEBUG[1224]: chan_sip.c:1309 create_addr: Setting NAT on RTP to 0 May 27 18:08:06 DEBUG[1224]: chan_sip.c:1487 sip_call: Outgoing Call for 7011 May 27 18:08:06 DEBUG[1224]: chan_sip.c:1620 update_user_counter: Call from user '7011' is 1 out of 0 -- Called 7011 May 27 18:08:08 DEBUG[1224]: chan_zap.c:3768 __zt_exception: Exception on 15, channel 1 May 27 18:08:08 DEBUG[1224]: chan_zap.c:3080 zt_handle_...

...Discussion Subject: [asterisk-users] FAX Hi all, When trying to send a FAX I got the following error: Executing [003228949469 at micho:1] Dial("SIP/028949469-08466918", "SIP/003228949469 at 80.169.210.181|60") in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949469 -- Couldn't call 0032234534534 at 1.1.1.1.1 Where I should define the codec other than the extension in order to succeed the call? Regards _______________________________________________ -- Bandwidth and Colocation Provide...

...IN IP4 79.80.x.x s=session c=IN IP4 79.80.x.x t=0 0 m=audio 16238 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jul 21 11:10:22] WARNING[23814]: chan_sip.c:2872 sip_call: Setting auto-congest time to 15000 ms. -- Called adf at 116.18.35.235:28614 <------------> ast-server*CLI> <--- SIP read from 116.18.35.235:28614 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678 Contact: <sip:adf at 116.18.35.235:2861...

...pp_dial.c:490 dial_exec: SIMPLE DIAL (NO URL) Dec 13 13:13:05 DEBUG[-1296254032]: chan_sip.c:2395 sip_alloc: Allocating new SIP call for (null) Dec 13 13:13:05 DEBUG[-1296254032]: chan_sip.c:1295 create_addr: Setting NAT on RTP to 0 Urgent handler Dec 13 13:13:05 DEBUG[-1296254032]: chan_sip.c:1536 sip_call: Outgoing Call for 2005 Dec 13 13:13:05 DEBUG[-1296254032]: chan_sip.c:1669 update_user_counter: Call from user '2005' is 1 out of 0 -- Called 2005 Urgent handler Dec 13 13:13:05 DEBUG[-1296254032]: chan_oh323.c:1136 oh323_indicate: OH323/R27015: Indicating condition 3. Dec 13 13:13:05...

...2 DEBUG[3242]: chan_zap.c:1384 zt_enable_ec: No echocancellation requested -- Executing Dial("Zap/3-1", "SIP/15111111111@SIP_PROVIDER") in new stack May 26 09:49:02 DEBUG[3242]: chan_sip.c:1309 create_addr: Setting NAT on RTP to 0 May 26 09:49:02 DEBUG[3242]: chan_sip.c:1487 sip_call: Outgoing Call for 15111111111 May 26 09:49:02 DEBUG[3242]: chan_sip.c:1592 update_user_counter: 15111111111 is not a local user -- Called 15111111111@SIP_PROVIDER May 26 09:49:03 DEBUG[3227]: chan_sip.c:822 __sip_ack: Acked pending invite 102 May 26 09:49:03 DEBUG[3227]: chan_sip.c:840 __sip_a...

All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and

...Discussion Subject: [asterisk-users] FAX Hi all, When trying to send a FAX I got the following error: Executing [003228949469 at micho:1] Dial("SIP/028949469-08466918", "SIP/003228949469 at 80.169.210.181|60") in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949469 -- Couldn't call 0032234534534 at 1.1.1.1.1 Where I should define the codec other than the extension in order to succeed the call? Regards _______________________________________________ -- Bandwidth and Colocation Provide...

I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all

...phone1' is 1 out of 0 DEBUG[1133735216]: File chan_sip.c, Line 3307 (build_route): build_route: Contact hop: <sip:phone1@10.1.2.24:5060;line=1> -- Executing Dial("SIP/phone1-3efc", "SIP/613@fwd.pulver.com") in new stack DEBUG[1209269552]: File chan_sip.c, Line 857 (sip_call): Outgoing Call for 613 DEBUG[1209269552]: File chan_sip.c, Line 952 (find_user): 613 is not a local user -- Called 613@fwd.pulver.com DEBUG[1133735216]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 548 (__sip_ack): Stopping retransmi...

...Discussion Subject: [asterisk-users] FAX Hi all, When trying to send a FAX I got the following error: Executing [003228949469 at micho:1] Dial("SIP/028949469-08466918", "SIP/003228949469 at 80.169.210.181|60") in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949469 -- Couldn't call 0032234534534 at 1.1.1.1.1 Where I should define the codec other than the extension in order to succeed the call? Regards _______________________________________________ -- Bandwidth and Colocation Provide...

...- Executing Macro("Zap/24-1", "stdexten|1312|SIP/1312") in new stack -- Executing Dial("Zap/24-1", "SIP/1312|20") in new stack Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1309 create_addr: Setting NAT on RTP to 0 Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1487 sip_call: Outgoing Call for 1312 Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1620 update_user_counter: Call from user '1312' is 1 out of 0 We're at asterisk.server.ip.addr port 19630 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with p...