similar to: Asterisk not recognizing ACK from an OK message? Help debuging SIP retransmit problem

I am using asterisk. I also have an access to VOIPSwitch ver 2 where I can see live calls. Many times I have seen that my calls are getting strucked and then it gets disconneected after 59 mins ( as settings are done accordingly in VOIPSwitch) What could be the reason ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have g723.1 codec in my asterisk. call leg from voipswitch is using codec g723.1 and call leg from gsm gateway is using codec gsm. I am having one way audio and getting below mentioned warning. Asterisk version is 1.8.11.0 [Jun 2 17:08:28] WARNING[21652]:

Hello, I'm fairly new to asterisk and Trixbox, I'm setting up a Trixbox based email to fax gateway. At this time, I have a ZAP PRI link between the eFax server and my VoIPSwitch. The ZAP channels are configured, the B and D channels are up, and I have green link lights on either end of my cabling, but when I dial the number I have assigned to my eFax server, the call never seems to route

Hello list, i need help. My problem is that when I want to call (using ISDN phone or internal SIP client) via the Sip provider a real phone number (ISDN phone or internal SIP Asterisk >> SIP ), I get a ring tone. When I now decide to hang up (e.g. if nobody answers), the called telephone continues to ring almost forever. the sip.conf: [2563105] accountcode = 2563105 username =

Hello, I am trying to add a sip-trunk to my Asterisk 1.4.15/Elastix 0.9.2 server. The system is in production with local extensions, a zap trunk and a working sip trunk with sipgate.de. My asterisk server is behind a NAT/Firewall, anyhow it registers and works well with sipgate.de on incoming and outgoing calls. I aquired an account with a reseller net-voz.com: I did some testing with the

Hi Friends, I want to use 5062 port for SIP protocol. I made the below modifications in my server to use 5062 port. Polycom phone: port=5062 Trunk settings: port=5062 sip.conf: bindaddr=5062 Extension configuration details: 5062 Our VoIP provider told me that they are allowing the SIP traffic through 5060 to 5064. I observed on my server console that my server is registered with our VoIP

Hello. I am in a strange situation. I have two asterisk. Asterisk "A" makes a call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers it to Openser by SIP. The problem is openser printing this in the screen: ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> . ERROR:parse_from_header: bad from header

Andrew Martin wrote: > ----- Original Message ----- <snip> > > By doing a number of test calls today, I have managed to reproduce this while > sip debugging was on, so I have that information available now as well: > http://pastebin.com/ZJqzdvY3 > > This was a call from 113 to 146 via a queue. Note that the asterisk server is > at 10.10.32.251. I see the following:

Hello, I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All the SIP clients are on a LAN, so no NAT is involved. I have been experiencing an intermittent problem where a call will be successfully answered, but then dropped by Asterisk 32 seconds after it is answered (with a "Retransmission timeout reached on transmission" error). Here is an example of this

Hi, I seem to be facing an intrusion issue, inspite of firewall (script attached). What am I missing ?? Any suggestions / recommendation are welcome pls. Best regards, Sans -------------- next part -------------- #!/bin/bash echo 0 > /proc/sys/net/ipv4/ip_forward # Clear any existing firewall stuff before we start /sbin/iptables --flush # As the default policies, drop all incoming

I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip

Greetings I have been running * for about a month now. Configuration. (5) Cisco 79xx IP phones (1) XP100P Pentium III (300mhz) 192meg memory Redat 8.0 (updated) It seems to run for about 3-6 hours, then the process stops. I have noticed, that * does not stop, if I do NOT have it register to other sip servers. (FWD and PCH). Here is are the last few lines in the /var/log/asterisk/messages

Hi all! Am I missing some configuration or is it simply a bug: If Asterisk chan_sip is configured with bindport=5062, the port is missing on the outgoing SIP messages contact header. This resulting in in-dialog messages sent to port 5060 ... where there is no Asterisk on that host... Tried externip = 1.2.3.4:5062 with no success. Version 1.6.0.3. br Walter -------------- next part

Anyone have any testing experience with T38 and HT-502 Grandstream? I just want to confirm that t.38 is working on this device. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110801/fea18d8c/attachment.htm>

I'm trying to set up pjsip to work with an obi202 and google voice. But I can't configure the endpoint. pjsip: [obi202-auth](!) type = auth auth_type = userpass password = <mypass> [obi202-aor](!) type = aor max_contacts = 2 ; ===== endpoints ======== [gv-voice](obi202-endpoint) auth = gv-voice aors = gv-voice identify_by=auth_username ;identify_by=username ; I also tried

Hello, Hope somebody can help me - Asterisk is behaving very oddly and I'm totally stumped! I have SER and Asterisk running on the same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will purely act as a voicemail server at the moment. However I cannot get Asterisk to listen on a

Hi Mark, Current CVS, * isn't correctly remembering the tag added to the To header by a server. For instance: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 81.96.69.210:5062;branch=7b858c09 From: steve-ata186 <sip:asterisk@81.96.69.210:5062>;tag=14925711 To: <sip:18478974611@4.42.235.170>;tag=t2907cab0911c8g Call-ID: 232752ec518d398f25ce03f45a8940f3@81.96.69.210 CSeq: 102 INVITE

Hi All, I'm trying to connect to a SIP carrier who never connected with Asterisk. I managed to connect with a sipura phone or a grandstream, no problem. When I configure asterisk, I'm able to send out calls to the carrier no problems, however, receiving calls doesn't work, and I keep getting the following messages: <-- SIP read from 69.xx.xx.xx:5060: INVITE

Calling my home numbers has always worked. Till now. The Asterisk CLI show the following : [Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to '<sip:092xx90xx at 85.xx.xx.xx>;tag=as5b139383' And after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of

Hello, For lab testing, I'm trying to build two differents PJSIP trunks between two Asterisk 13.8.0enabled boxes. I thought I could set up both trunks like this: Box A/port 5060 <------ Trunk1 -----> Box B/port 5060 Box A/port 5062 <------ Trunk2 -----> Box B/port 5062 and declare trunks like this: [foobar1] type=endpoint transport=simpletrans context=from-customer